Dtmf 500 server internal error

dtmf 500 server internal error

How fast is your download speed? In seconds, FAST.com's simple internet speed test will estimate your ISP speed. Fixed device does not register if DNS server responds without a Fixed Broadsoft Interop: device returns 500 Internal Error to MWI NOTIFY. 10500; reason="Gateway responded with 500 Server Internal Error"; PI2IP=8 SigIPDF=40 CNGMode=0 DTMFUsed=0 NSEMode=0 PlayRBTone2IP=0.

Dtmf 500 server internal error - think, that

StarTrinity.com

SIP trace is a series of captured/recorded SIP packets, SIP packet data represented as text. The SIP trace is necessary element for VoIP troubleshooting. When something goes wrong with your phone system, the first thing to look at is SIP trace. SIP trace and pcap recordings provide evidence of VoIP issues.

SIP messages contain request/status line (red), headers (blue) and body (green) (format of body is defined by protocol SDP). Here is sample SIP trace:
SIP messages for Call-ID 524c3ea2fdd1401ab4a90c940b073359

17:42:22.442 [+0,00ms] [TX] INVITE to 5.135.179.50:6000
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.4:5090;rport;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Max-Forwards: 70
From: sip:[email protected];tag=d03ca0a253134ecc82c17a89e4b31e45
To: sip:[email protected]
Contact: <sip:[email protected]:5090>
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
CSeq: 17970 INVITE
Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Supported: 100rel, timer
User-Agent: StarTrinity.SIP 2018-06-14 19.29 UTC
Session-Expires: 72000;refresher=uas
Content-Type: application/sdp
Content-Length: 291

v=0
o=- 3739369344 3739369344 IN IP4 192.168.10.4
s=o3.proxy.stream0
c=IN IP4 192.168.10.4
t=0 0
m=audio 21504 RTP/AVP 8 0 4 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

============================================================end of message=================
17:42:22.532 [+90,26ms] [RX] Trying from 5.135.179.50:6000
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.4:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
From: <sip:[email protected]>;tag=d03ca0a253134ecc82c17a89e4b31e45
To: <sip:[email protected]>
CSeq: 17970 INVITE
Supported: 100rel, timer
Content-Length: 0


============================================================end of message=================
17:42:22.542 [+100,16ms] [RX] Session Progress from 5.135.179.50:6000
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.4:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
From: <sip:[email protected]>;tag=d03ca0a253134ecc82c17a89e4b31e45
To: <sip:[email protected]>;tag=e5316920762f4d01b4618882a3fbcad4
CSeq: 17970 INVITE
Supported: 100rel, timer
Contact: <sip:[email protected]:6000>
Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Server:
Content-Type: application/sdp
Content-Length: 251

v=0
o=- 3739358524 3739358524 IN IP4 5.135.179.50
s=i2.proxy.stream0
c=IN IP4 5.135.179.50
t=0 0
m=audio 26000 RTP/AVP 8 101
a=rtcp:26001 IN IP4 5.135.179.50
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

============================================================end of message=================
17:42:25.584 [+3 142,67ms] [RX] Service Unavailable from 5.135.179.50:6000
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.10.4:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
From: <sip:[email protected]>;tag=d03ca0a253134ecc82c17a89e4b31e45
To: <sip:[email protected]>;tag=e5316920762f4d01b4618882a3fbcad4
CSeq: 17970 INVITE
Supported: 100rel, timer
Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Server:
Content-Length: 0


============================================================end of message=================
17:42:25.584 [+3 142,70ms] [TX] ACK to 5.135.179.50:6000
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.4:5090;rport;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Max-Forwards: 70
From: sip:[email protected];tag=d03ca0a253134ecc82c17a89e4b31e45
To: sip:[email protected];tag=e5316920762f4d01b4618882a3fbcad4
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
CSeq: 17970 ACK
Content-Length: 0


============================================================end of message=================
===============================saved by StarTrinity SIP Tester at 2018-06-30 18:27:45======
You can see INVITE request sent by UAC, multiple responses sent by UAS (100 Trying, 183 Session Progress, 503 Service Unavailable). SIP Request LineINVITE sip:[email protected] SIP/2.0 means request type = "INVITE" (making a VoIP call), SIP protocol version = "2.0" (the only one version used, as we see in 2018), and destination SIP URI = "sip:[email protected]". The URI in SIP is simular to URL in HTTP, it specifies target for the request. "sip:" or "sips:" here define protocol type, "123456" is called number (telephone number of B endpoint).
In next chapters we will describe the SIP trace in detail.
Types of SIP packets
SIP Requests also known as SIP methods, SIP request methods. Most popular SIP request types:
  • INVITE - most widely used SIP request, means request to make new VoIP call from UAC to UAS. When the INVITE is sent after connection (after "200 OK response") it usually means change of audio codec, putting call on hold, or sending/receiving fax
  • ACK - confirmation that response was delivered to UAS
  • BYE - ends current VoIP call after answer, sent by UAS or UAC
  • CANCEL - ends a call before answer, sent by UAC
  • REGISTER - sent from SIP phone user-agent to SIP server to register the phone to specific extension (i.e. internal phone number)
  • REFER - starts call transfer procedure: e.g. A makes call, B answers, understands that call needs to be passed to C and sends request to B: "please transfer to C"
  • We have listed only most popular SIP request types, see others in RFC3261
SIP Responses
The SIP response packets are is sent back to requesting endpoint and contain SIP response code and optional error details in SIP headers:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 9.45.33.11:5081;branch=z9hG4bK-10854-1-0
From: sipp <sip:[email protected]:5081>;tag=10854SIPpTag001
To: sut <sip:[email protected]:5060>;tag=38404-23F2
Date: Tue, 05 Feb 2013 05:49:37 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Warning: 399 9.43.29.50 "Transcoder Not Configured"
Server: Cisco-SIPGateway/IOS-15.3.20130202.123643.
Reason: Q.850;cause=16
Content-Length: 0
In this example "Warning" header gives details about the error and "Reason" header gives information about PSTN release code.
If a packet is lost somewhere between the endpoints A or B (UDP protocol does not gaurantee delivery of packets), the SIP packets are retransmitted.
Task: see types of SIP packets (requests, responses), and number of retransmitted packets in file #1, file #2, file #3. See what is destination IP address where packets are retransmitted to, do you think it is correct or not?
SIP headers
A header in SIP packet has a simple format: [name]: [value]. SIP headers are separated by line breaks. Most widely used SIP headers are
  • From header indicates identity of caller, usually contains Caller ID (ANI, CLI, A number). Also contains unique "tag" set by UAC
  • To header identifies recipient of the call, usually contains Called ID (called number, B number). Also contains unique "tag" set by UAS
  • Via header saves information about proxy servers - server IP address and port - to pass response from UAS back to UAC via same set of proxy servers
  • Record-Route header is optionally inserted by proxy servers to route subsequent SIP requests for the same SIP call
  • Call-ID header identifies SIP call along with "From" and "To" headers
  • Contact header specifies IP address and port to be used by peer SIP endpoints for subsequent SIP requests for the same SIP call
  • CSeq header contains a decimal number which increases by 1 with every subsequent SIP request within the same SIP call, with the exception of CANCEL and ACK requests which use the CSeq number of corresponding INVITE request
  • Server header indicates model, software/firmware version of UAS
  • User-Agent header indicates model, software/firmware version of UAC
  • Content-Length header as you will read later, INVITE and "200 OK" packets usually contain encapsulatedSDP protocol data with information about RTP audio streams. This header specifies size of encapsulated SDP (or other) data
  • Content-Type header specifies type of encapsulated data. Equals to "application/sdp" in case of encapsulated SDP
  • Allow header lists set of SIP requests supported by user agent which generated the SIP message
  • Supported header specifies set of supported SIP extensions by user agent which generated the SIP message
  • Require header specifies set of required SIP extensions by user agent which generated the SIP message
  • Session-Expires header - part of RFC 4028 (Session Timers) SIP extension - specifies max. interval to refresh SIP call with keep-alive packet (the RFC3261 standard does not define way to abort hang calls)
  • Expires header specifies lifetime of corresponding SIP request
  • WWW-Authenticate, Authorization headers are used for digest authentication
There are many more headers; adding custom headers into requests and responses is a way to extend SIP protocol.
Tasks:
  • See SIP headers in this file: what software is used for UAC and UAS? What is SIP Call ID?
  • See how to fix Contact header sent by UAS in this file
Registration
Registration in SIP means passing information about user agent to server, it is implemented with REGISTER. It means "Hello SIP server, here are my contact details for extension number 456: IP address 1.2.3.4, SIP port 5060". Without registration SIP phones must know IP addresses of every destination phone; with registration the IP addresses are stored by SIP server, also known as SIP registrar server. Her is example SIP trace for the registration process:
SIP messages for Call-ID 64b10daa51dd4c5a9d6038a43163b22d

14:32:00.599 [+0,00ms] [TX] REGISTER to 5.135.179.50:6000
REGISTER sip:startrinity.com:6000 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.4:5090;rport;branch=z9hG4bKPj609d6772ba754dcab9cba92bcb34be69
Max-Forwards: 70
From: <sip:[email protected]>;tag=aef6297c37914f69832efc2c44c980aa
To: <sip:100@startrinity.com>
Call-ID: 64b10daa51dd4c5a9d6038a43163b22d
CSeq: 22151 REGISTER
Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Contact: <sip:[email protected]:5090>
User-Agent: StarTrinity.SIP 2018-06-14 19.29 UTC
Expires: 3600
Content-Length: 0


============================================================end of message=================
14:32:00.674 [+74,84ms] [RX] Unauthorized from 5.135.179.50:6000
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.117.173.127:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPj609d6772ba754dcab9cba92bcb34be69
Call-ID: 64b10daa51dd4c5a9d6038a43163b22d
From: <sip:[email protected]>;tag=aef6297c37914f69832efc2c44c980aa
To: <sip:[email protected]>
CSeq: 22151 REGISTER
WWW-Authenticate: Digest realm="*",nonce="6ab702ae1b742592",opaque="66982f5173f64308",algorithm=md5
Content-Length: 0


============================================================end of message=================
14:32:00.674 [+74,88ms] [TX] REGISTER to 5.135.179.50:6000
REGISTER sip:startrinity.com:6000 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.4:5090;rport;branch=z9hG4bKPjcfb02f366c7f435ba7991cbb13b9497c
Max-Forwards: 70
From: <sip:[email protected]>;tag=aef6297c37914f69832efc2c44c980aa
To: <sip:[email protected]>
Call-ID: 64b10daa51dd4c5a9d6038a43163b22d
CSeq: 22152 REGISTER
Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Contact: <sip:[email protected]:5090>
User-Agent: StarTrinity.SIP 2018-06-14 19.29 UTC
Expires: 3600
Authorization: Digest username="100", realm="*", nonce="6ab702ae1b742592", uri="sip:startrinity.com:6000", response="abd79c866205325d0511a95b1e5f5087", algorithm=md5, opaque="66982f5173f64308"
Content-Length: 0


============================================================end of message=================
14:32:00.758 [+158,50ms] [RX] OK from 5.135.179.50:6000
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.117.173.127:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPjcfb02f366c7f435ba7991cbb13b9497c
Call-ID: 64b10daa51dd4c5a9d6038a43163b22d
From: <sip:[email protected]>;tag=aef6297c37914f69832efc2c44c980aa
To: <sip:[email protected]>
CSeq: 22152 REGISTER
Contact: sip:[email protected]:5090;expires=3600
Content-Length: 0


============================================================end of message=================
===============================saved by StarTrinity SIP Tester at 2018-07-02 14:32:12======
As you can see in the SIP trace, first REGISTER request sent by UAC 192.168.10.4 to UAS 5.135.179.50 means resitration of extension 100 at SIP server startrinity.com on SIP port 6000. The UAC sends its IP address in Contact header: Contact: <sip:[email protected]:5090>. "Expires" SIP header equals to 3600 means that the registration entry should be valid for 3600 seconds - in 60 minutes the registrar server should clear the registration entry if not received new REGISTER requests for this extension. REGISTER request with "Expires: 0" means unregistration.
Digest authentication
As you can see in previous SIP trace (with REGISTER), first request is rejected with status "401 Unauthorized" and WWW-Authenticate header, it is called digest challenge. The WWW-Authenticate SIP header contains nonce and opaque values which are used by UAC along with user name and password to send second INVITE request. The second INVITE request contains Authorization header with digest response, it is used by UAS to verify user name and password. If user name and password are correct, UAS sends "200 OK" response, otherwise it sends "403 Unauthorized" response.
SIP extensions
Extensions of SIP protocol are defined by additional RFC standard documents (beyond RFC3261), you can find them here. Most important SIP extensions are
  • 100rel extension defines PRACK method to provide reliable delivery of provisionalSIP responses. It is fully described in RFC3262.
    The PRACK packets are sent in response to 18x provisional responses via chain of SIP proxy servers from UAC to UAS:
    PRACK packets going via SIP proxies
    Here is example SIP call flow for PRACK:
    PRACK call flow
    Without the PRACK, 18x SIP responses could be lost in IP network between UAS and UAC, it means no ringback tone played to caller. With PRACK, 18x SIP response is retransmitted by UAS if it does not receive PRACK from UAC.
  • timer extension defines a keep-alive mechanism for SIP calls, it is described in RFC4068 (Session Timers). The keep-alive packets are sent as re-INVITE or UPDATE SIP packets between UAC and UAS; if the keep-alive packet is not received within session interval, UAC or UAS considers call as hang (dead) and drops the call. Without the session timer extension there could appear hang calls, existing on one SIP UA and already destroyed on another SIP UA. Such situation can happen if internet connection drops between UAC and UAS during end of call, and BYE packet gets lost.

Call transfer

The VX Integrated Media Platform provides the following types of call transfers:

Note: In this topic, the A leg refers to the initial call made to or from the VX Integrated Media Platform.

Bridge transfer

A bridge transfer occurs when the VX Integrated Media Platform initiates a transfer between two parties, and the caller returns to the VX Integrated Media Platform after the transfer ends (when the caller disconnects from the third party). With a bridge transfer:

  • The VX Integrated Media Platform is aware of the outcome of the transfer.

  • The original caller is not disconnected in the event of a connection error.

A bridge transfer appears as a new SIP INVITE from the VX Integrated Media Platform. The audio mixing occurs in the Media Resource Function component of the VoiceXML Interpreter. This component also performs DTMF hotword detection on the A leg.

The dest attribute has different meanings, depending on whether the call is being transferred to a SIP target or a PSTN target:

For this transfer target...

The dest attribute specifies the...

SIP

Destination SIP URI of the transfer target.

Place this SIP call by using the com.vision.miosip.rvsip.outboundProxyHost configuration setting.

PSTN

User part of the To SIP URI.

The remainder of the URI is constructed using the URI from the com.vision.miosip.dialog.TelHosts configuration setting.

The dest attribute is specified using the tel: URI syntax and can include the custom parameters connecttimeout and maxtime. A tel: URI is converted to a SIP URI, with the user=phone parameter indicating that the call must be placed over the PSTN network.


The following example shows a tel: URI, along with its custom parameters:

tel:12345678;maxtime=60s&connecttimeout=10s.

The following table describes the custom parameters:

Custom parameter

Description

connecttimeout

Amount of time to wait for a final SIP response to the INVITE. When this time limit is reached, a CANCEL is issued and the transfer aborted. The timer for the connecttimeout starts when the 180 Ringing response is received.

maxtime

Maximum duration allowed for a call. When this time limit is reached, a BYE is issued to the outbound leg. The SIP INVITE contains the custom header Vision-ParentCallID, which equates to the Call-ID of the A leg.


While the outbound call is being attempted, the A leg hears the audio specified by the com.vision.miosip.defaultTransferAudio configuration setting.

The following table describes the mapping of SIP responses from the INVITE to the value of either the transfer form item variable or the VoiceXML event:

SIP response

Transfer form item variable / VoiceXML event

404 Not Found

error.connection.baddestination

408 Request Timeout

noanswer

486 Busy Here

busy

500 Server Internal Error

unknown

503 Service Unavailable

noresource

603 Decline

noanswer


The following table describes how the transfer form item variable is set when the outbound leg of a call is terminated:

Action

Transfer form item variable setting

If the called party hangs up (issues a BYE), the outbound call terminates.

far_end_disconnect

VoiceXML Interpreter terminates the outbound leg because the maximum call duration expired or because a hotword was detected.

near_end_disconnect


Blind transfer

A blind transfer occurs when the VX Integrated Media Platform initiates a transfer between two parties and then detaches from the call before the transfer takes place. With a blind transfer:

  • The VX Integrated Media Platform is not aware of the outcome of the transfer.

  • The original caller is disconnected if there is a transfer error.

A blind transfer uses the SIP REFER method (RFC 3515). After a successful response to the REFER message, the VX Integrated Media Platform generates a connection.disconnect.transfer and issues a BYE to drop the call.

The following table describes the attributes of the transfer element:

Attribute

Description

dest

Sets the user part of the SIP URI specified by the Refer-To header in the REFER message.

aai

Sets the aai part of the SIP URI specified by the Refer-To header in the REFER message.


The following example shows the use of a REFER message:

REFER sip:[email protected]:5060 SIP/2.0
From: <sip:[email protected]>
To: <sip:10.3.1.52>;tag=ds-42d6-bb2c
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 1 REFER
Content-Length: 0
Refer-To: sip:[email protected];connecttimeout=2s;maxtime=60s
Via: SIP/2.0/UDP 10.0.0.99:5060

Valid responses to the REFER message are:

  • 202 Accepted

  • 500 Server Internal Error if an error occurs

Consultation transfer

A consultation transfer occurs when the VX Integrated Media Platform initiates a transfer between two parties and then detaches from the call after the transfer takes place. With a consultation transfer:

  • The VX Integrated Media Platform is aware of the outcome of the transfer.

  • The original caller is not disconnected in the event of a connection error.

A consultation transfer is also called a supervised transfer.

The following table describes the consultation transfer process:

Task

Description

1

The VX Integrated Media Platform creates a new call leg by issuing an INVITE message to the outbound server.

2

If the server accepts the call, the VX Integrated Media Platform issues a REFER message on the new call leg.

The Refer-To header contains a Replaces field that holds the Call ID value for the new leg. This causes the new call leg to be replaced by the initial call leg on the remote UAC.

3

Control returns to the VoiceXML context.

4

A <connection.disconnect.transfer> event is issued.


[cisco-voip] SIP 500 Error on CUBE and CUCMBE

Jason Aarons (AM)jason.aarons at dimensiondata.com
Tue Sep 3 11:29:17 EDT 2013
See; m=audio 19426 RTP/AVP 18 101 The 18 equals G729, the 101 is related to DTMF. You should see AVP 0 101 for G711. Watch your debug ccsip messages for the codecs being sent in the SDP. It looks like you have a mismatch, I wasn't clear the direction of the below SIP packet (inbound vs outbound) but it's not showing G711 as you indicated is your codec. Which would requires a Transcoder. https://supportforums.cisco.com/docs/DOC-27105 From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kenneth Hayes Sent: Sunday, September 01, 2013 10:57 AM To: Cisco VoIPoE List Subject: [cisco-voip] SIP 500 Error on CUBE and CUCMBE I posted this on the Cisco forums but I wanted to reach out to you guys here as well. So Inbound calls works fine with no issues, however outbound I'm getting a busy signal when I dial out to the PSTN, not your typical CSS fast busy but busy like someone who does not have call waiting. Here are the debugs below. *Sep 1 15:33:16.855: //10/38360D000000/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK12d271b8c5e From: "user - 1000" <sip:2089013875 at 192.168.1.8>;tag=645~82127680-9903-4752-ad60-3545686b7a7f-25789895 To: <sip:+17705000140 at 192.168.1.3>;tag=1DD304-1CA2 Date: Sun, 01 Sep 2013 15:33:16 GMT Call-ID: 38360d00-2231532b-11c-801a8c0 at 192.168.1.8<mailto:38360d00-2231532b-11c-801a8c0 at 192.168.1.8> CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Reason: Q.850;cause=44 Content-Length: 0 192.168.1.3= CUBE 192.168.1.8= CUCMBE v=0 o=CiscoSystemsCCM-SIP 645 1 IN IP4 192.168.1.8 s=SIP Call c=IN IP4 192.168.1.3 t=0 0 m=audio 19426 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 SIP trunk between CUCMBE and CUBE are MTP codec method is 711ulaw, in my Regions my CUBE is set to g729r8, and phones are set to 711ulaw. I'm not sure what's happening but it's very frustrating. Anyone can be of assistance? _______________________________________________ cisco-voip mailing list cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net> https://puck.nether.net/mailman/listinfo/cisco-voip itevomcid -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20130903/a5321bb8/attachment.html>

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I definitely got somewhere today. I was able to get the credentials for the SIP paging setup:

Destination Address: (IP Of Server)
Destination Port: 5060
Transport Type: UDP
Security: None
Default Telephony Event Payload Type: 101
DTMF Signaling Method: RFC2833 or SIP INFO

My SIP Trunk PEER info looks like this:

type=peer
host=(IP Address)
port=5060
transport=udp
dtmfmode=rfc2833
insecure=port,invite

Under Outbound Routes, I have a route set up for the trunk. Most of the options are default, but I did set:

Route Settings/Trunk Sequence for Matched Routes: Lists the SIP Trunk I created
Dial Patterns: Has 19600 in the prefix field, as that is the extension to dial to page

When I get on one of the phones and dial 19600, I get “All circuits are busy.”
When I do a verbose logging, I see many things, but the most telling is this:

== Spawn extension (from-trunk-sip-TrunkToSynApps, 19600, 1) exited non-zero on ‘SIP/TrunkToSynApps-00000011’
– SIP/TrunkToSynApps-00000011 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called SIP/TrunkToSynApps/
– Got SIP response 500 “Server Internal Error” back from (IP Address):5060
– SIP/TrunkToSynApps-00000011 is circuit-busy

Based on the information I have provided, is there an error in my PEER configuration, or is this something that is a problem with the server?

In the following diagram below, an H.323 endpoint (GW or GK) initiates a session by sending a Setup request destined for a SIP endpoint (such as a UA or a SIP Gateway). Between these entities, the Oracle® Enterprise Session Border Controller is positioned to perform interworking. The H.323 endpoint has completed the RAS process prior to sending the SETUP message.

The call flow for this type of translation works fundamentally the same way that the translation does for H.323 Fast Start to SIP, with the exception of how the media is established. When the Oracle® Enterprise Session Border Controller receives an H.323 message destined for a SIP endpoint, it sends a SIP INVITE message that includes default SDP to that SIP endpoint. The default SDP is constructed using information in the media profiles listed for the IWF configuration; if necessary, this media information is amended later in the sequence. Once the call is set up, the Oracle® Enterprise Session Border Controller negotiates media with the H.323 endpoint through a series of TCS/TCS Ack and OLC/OLC Ack messages that establish the operating mode and Gateway capability.

When the Oracle® Enterprise Session Border Controller completes media negotiation with the H.323 endpoint, it issues a re-INVITE to the SIP endpoint that contains the updated information needed for media transmission. In response, the SIP endpoint sends a 200 OK message that the Oracle® Enterprise Session Border Controller answers with an ACK. Then RTP can flow between the two endpoints.

The H.323 Slow Start to SIP call flow is described above.

Troubleshooting

Overview

The Vonage Voice API offers a highly available service. However, due to the nature of providing service across hundreds of phone carriers around the world, problems may arise occasionally that are outside of Vonage's control. In addition there are certain limitations placed on Vonage by partner networks, which can have an impact on how your application functions. While not exhaustive here are a few things to look for if you are experiencing problems.

Timeouts

When Vonage sends a webhook to your Answer URL it expects the server to respond in a certain time frame:

  1. The TCP connection should be established within 3 seconds.
  2. The HTTP response (NCCO) should be returned within 5 seconds.

If Vonage does not get a response within these time frames it will retry the request twice. If this fails, Vonage will make two further attempts to access your Fallback Answer URL, if it is configured. If Vonage does not get a response within these time frames from your Fallback Answer URL it will retry again. If the Fallback Answer URL responds with a HTTP error code or invalid NCCO then the call is disconnected.

Regions

The Vonage Voice API resides in five geographic data centers. Phone numbers are associated with the closest data center: US East Coast, Texas, London, Amsterdam or Singapore. API requests are routed to the closest data center to the requesting client. However, a call currently only exists in a single region, this means that if you are receiving a call on a number connected to Singapore but making an API request from a server hosted in the US it will return a 404.

You can work around this issue by sending your API request to the correct region, either:

  • https://api-us-1.nexmo.com (Washington DC)
  • https://api-us-2.nexmo.com (Dallas)
  • https://api-eu-1.nexmo.com (London)
  • https://api-eu-2.nexmo.com (Amsterdam)
  • https://api-sg-1.nexmo.com (Singapore)

API endpoint corresponding to particular call is returned as parameter in Answer webhook.

The following are examples of how to override the default hosts using the Vonage Server SDKs:

Capacity

As standard the Voice API has the following capacity limitations:

  1. Maximum of 3 outgoing calls per second created either via the REST API or using the action within another call.
  2. Maximum of 15 requests per second to the REST API (excluding create calls).

If you exceed these limits you will receive an HTTP 429 response or a webhook to your with an error.

Error events

The Vonage Voice API sends error events to the associated with the application. For example, when we encounter an invalid NCCO or an outbound call failure.

Consider, what: Dtmf 500 server internal error

Dtmf 500 server internal error
Mbr error 2
ERROR UDEVD FAILED TO STOP
Dtmf 500 server internal error
FFMPEG HTTP ERROR 400 BAD REQUEST
dtmf 500 server internal error

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Fix 500 Internal Server Error android

In the following diagram below, an H.323 endpoint (GW or GK) initiates a session by sending a Setup request destined for a SIP endpoint (such as a UA or a SIP Gateway). Between these entities, the Oracle® Enterprise Session Border Controller is positioned to perform interworking, dtmf 500 server internal error. The H.323 endpoint has completed the RAS process prior to sending the SETUP message.

The call flow for this type of translation works fundamentally the same way that the translation does for H.323 Fast Start to SIP, with the exception of how the media is established. When the Oracle® Enterprise Session Border Controller receives an H.323 message destined for a SIP endpoint, it sends a SIP INVITE message that includes default SDP to that SIP endpoint. The default SDP is constructed using information in the media profiles listed for the IWF configuration; if necessary, this media information is amended later in the sequence. Once the call is set up, the Oracle® Enterprise Session Border Controller negotiates media with the H.323 endpoint through a series of TCS/TCS Ack and OLC/OLC Ack messages that establish the operating mode and Gateway capability.

When the Oracle® Enterprise Session Border Controller completes media negotiation with the H.323 endpoint, it issues a re-INVITE to the SIP endpoint that contains the updated information needed for media transmission. In response, the SIP endpoint sends a 200 OK message that the Oracle® Enterprise Session Border Controller answers with an ACK. Then RTP can flow between the two endpoints.

The H.323 Slow Start to SIP call flow is described above.

Troubleshooting

Overview

The Vonage Voice API offers a highly available service. However, due to the nature of providing service across hundreds of phone carriers around the world, problems may arise occasionally that are outside of Vonage's control. In addition there are certain limitations placed on Vonage by partner networks, which can have an impact on how your application functions. While not exhaustive here are a few things to look for if you are experiencing problems.

Timeouts

When Vonage sends a webhook to your Answer URL it expects the server to respond in a certain time frame:

  1. The TCP connection should be established within 3 seconds.
  2. The HTTP response (NCCO) should be returned within 5 seconds.

If Vonage does not get a response within these time frames it will retry the request twice. If this fails, Vonage will make two further attempts to access your Fallback Answer URL, if it is configured. If Vonage does not get a response within these time frames from your Fallback Answer URL it will retry again. If the Fallback Answer URL responds with a HTTP error code or invalid NCCO then the call is disconnected.

Regions

The Vonage Voice API resides in five geographic data centers. Phone bde error capability not supported are associated with the closest data center: US East Coast, Texas, London, Amsterdam or Singapore, dtmf 500 server internal error. API requests are routed to the closest data center to the requesting client. However, a call currently only exists in a single region, this means that if you are receiving a call on a number connected to Singapore but making an API request from a server hosted in the US it will return a 404.

You can work around this issue by sending your API request to the correct region, either:

  • https://api-us-1.nexmo.com (Washington DC)
  • https://api-us-2.nexmo.com (Dallas)
  • https://api-eu-1.nexmo.com (London)
  • https://api-eu-2.nexmo.com (Amsterdam)
  • https://api-sg-1.nexmo.com (Singapore)

API endpoint corresponding to particular call is returned as parameter in Answer webhook.

The following are examples of how to override the default hosts using the Vonage Server SDKs:

Capacity

As standard the Voice API has the following capacity limitations:

  1. Maximum of 3 outgoing calls per second created either dtmf 500 server internal error the REST API or using the action within another call.
  2. Maximum of 15 requests per second to the REST API (excluding create calls).

If you exceed these limits you will receive an HTTP 429 response or a webhook to your with an error.

Error events

The Vonage Voice API dtmf 500 server internal error error events to the associated with the application. For example, when we encounter an invalid NCCO or an outbound call failure.

FreePBX Community Forums

I definitely got somewhere today. I was able to get the credentials for the SIP paging setup:

Destination Address: (IP Of Server)
Destination Port: 5060
Transport Type: UDP
Security: None
Default Telephony Event Payload Type: 101
DTMF Signaling Method: RFC2833 or SIP INFO

My SIP Trunk PEER info looks like this:

type=peer
host=(IP Address)
port=5060
transport=udp
dtmfmode=rfc2833
insecure=port,invite

Under Outbound Routes, I have a route set up for the trunk. Most of the options are default, but I did set:

Route Settings/Trunk Sequence for Matched Routes: Lists the SIP Trunk I created
Dial Patterns: Has 19600 in the prefix field, as that is the extension to dial to page

When I get on one of the phones and dial 19600, I get “All circuits are busy.”
When I do a verbose logging, I see many things, but the most telling is this:

== Spawn extension (from-trunk-sip-TrunkToSynApps, 19600, 1) exited non-zero on ‘SIP/TrunkToSynApps-00000011’
– SIP/TrunkToSynApps-00000011 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called SIP/TrunkToSynApps/
– Got SIP response 500 “Server Internal Error” back from (IP Address):5060
– SIP/TrunkToSynApps-00000011 is circuit-busy

Based on the information I have provided, is there an error in my PEER configuration, or is this something that is a problem with the server?

StarTrinity.com

SIP trace is a series of captured/recorded SIP packets, SIP packet data represented as text. The SIP trace is necessary element for VoIP troubleshooting. When something goes wrong with your phone system, the first thing to look at is SIP trace. SIP trace and pcap recordings provide evidence of VoIP issues.

SIP messages contain request/status line (red), headers (blue) and body (green) (format of body is defined by protocol SDP). Here is sample SIP trace: bootcfg rebuild error failed to add
SIP messages for Call-ID 524c3ea2fdd1401ab4a90c940b073359

17:42:22.442 [+0,00ms] [TX] INVITE to 5.135.179.50:6000
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.4:5090;rport;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Max-Forwards: 70
From: sip:[email protected];tag=d03ca0a253134ecc82c17a89e4b31e45
To: sip:[email protected]
Contact: <sip:[email protected]:5090>
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
CSeq: 17970 INVITE
Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Supported: 100rel, timer
User-Agent: StarTrinity.SIP 2018-06-14 19.29 UTC
Session-Expires: 72000;refresher=uas
Content-Type: application/sdp
Content-Length: 291

v=0
o=- 3739369344 3739369344 IN IP4 192.168.10.4
s=o3.proxy.stream0
c=IN IP4 192.168.10.4
t=0 0
m=audio 21504 RTP/AVP 8 0 4 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

============================================================end of message=================
17:42:22.532 [+90,26ms] [RX] Trying from 5.135.179.50:6000
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.4:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
From: <sip:[email protected]>;tag=d03ca0a253134ecc82c17a89e4b31e45
To: <sip:[email protected]>
CSeq: 17970 INVITE
Supported: 100rel, timer
Content-Length: 0


============================================================end of message=================
17:42:22.542 [+100,16ms] [RX] Session Progress from 5.135.179.50:6000
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.4:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
From: <sip:[email protected]>;tag=d03ca0a253134ecc82c17a89e4b31e45
To: <sip:[email protected]>;tag=e5316920762f4d01b4618882a3fbcad4
CSeq: 17970 INVITE
Supported: 100rel, timer
Contact: <sip:[email protected]:6000>
Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, dtmf 500 server internal error, CANCEL, UPDATE
Server:
Content-Type: application/sdp
Content-Length: 251

v=0
o=- 3739358524 3739358524 IN IP4 5.135.179.50
s=i2.proxy.stream0
c=IN IP4 5.135.179.50
t=0 0
m=audio 26000 RTP/AVP 8 101
a=rtcp:26001 IN IP4 5.135.179.50
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

============================================================end of message=================
17:42:25.584 [+3 142,67ms] [RX] Service Unavailable from 5.135.179.50:6000
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.10.4:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
From: <sip:[email protected]>;tag=d03ca0a253134ecc82c17a89e4b31e45
To: <sip:[email protected]>;tag=e5316920762f4d01b4618882a3fbcad4
CSeq: 17970 INVITE
Supported: 100rel, timer
Allow: INFO, PRACK, SUBSCRIBE, dtmf 500 server internal error, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Server:
Content-Length: 0


============================================================end of message=================
17:42:25.584 [+3 142,70ms] [TX] ACK to 5.135.179.50:6000
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.4:5090;rport;branch=z9hG4bKPjefcc1928b9d44bf68b202f5674f15b9c
Max-Forwards: 70
From: sip:[email protected];tag=d03ca0a253134ecc82c17a89e4b31e45
To: sip:[email protected];tag=e5316920762f4d01b4618882a3fbcad4
Call-ID: 524c3ea2fdd1401ab4a90c940b073359
CSeq: 17970 ACK
Content-Length: 0


============================================================end of message=================
===============================saved by StarTrinity SIP Dtmf 500 server internal error at 2018-06-30 18:27:45======
You can see INVITE request sent by UAC, multiple responses sent by UAS (100 Trying, 183 Session Progress, 503 Service Unavailable). SIP Request LineINVITE sip:[email protected] SIP/2.0 means request type = "INVITE" (making a VoIP call), SIP protocol version = "2.0" (the only one version used, as we see in 2018), and destination SIP URI = "sip:[email protected]". The URI in SIP is simular to URL in HTTP, it specifies target for the request. "sip:" or "sips:" here define protocol type, "123456" is called number (telephone number of B endpoint).
In next chapters we will describe the SIP trace in detail.
Types of SIP packets
SIP Requests also known as SIP methods, SIP request methods. Most popular SIP request types:
  • INVITE - most widely used SIP request, means request to make new VoIP call from UAC to UAS, dtmf 500 server internal error. When the INVITE is sent after connection (after "200 OK response") it usually means change of audio codec, putting call on hold, or sending/receiving fax
  • ACK - confirmation that response was delivered to UAS
  • BYE - ends current VoIP call after answer, sent by UAS or UAC
  • CANCEL - ends a call before answer, sent by UAC
  • REGISTER - sent from SIP phone user-agent to SIP server to register the phone to specific extension (i.e. internal phone number)
  • REFER - starts call transfer procedure: e.g. A makes call, B answers, understands that call needs to be passed to C and sends request to B: "please transfer to C"
  • We have listed only dtmf 500 server internal error popular SIP request types, see others in RFC3261
SIP Responses
The SIP response packets are is sent back to requesting endpoint and contain SIP response code and optional error details in SIP headers:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 9.45.33.11:5081;branch=z9hG4bK-10854-1-0
From: sipp <sip:[email protected]:5081>;tag=10854SIPpTag001
To: sut <sip:[email protected]:5060>;tag=38404-23F2
Date: Tue, 05 Feb 2013 05:49:37 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Warning: 399 9.43.29.50 "Transcoder Not Configured"
Server: Cisco-SIPGateway/IOS-15.3.20130202.123643.
Reason: Q.850;cause=16
Content-Length: 0
In this example "Warning" header gives details about the error swf undefined error "Reason" header gives information about PSTN release code. ati2dvag error blue screen
If a packet is lost somewhere between the endpoints A or B (UDP protocol does not gaurantee delivery of packets), the SIP packets are retransmitted.
Task: see types of SIP packets (requests, responses), and number of retransmitted packets in file #1, file #2, file #3, dtmf 500 server internal error. See what is destination IP address where packets are retransmitted to, do you think it is correct or not?
SIP headers
A header in SIP packet has a simple format: [name]: [value], dtmf 500 server internal error. SIP headers are separated by line breaks. Most widely used SIP headers are islamist terrorism in the sahel pdf
  • From header indicates identity of caller, usually contains Caller ID (ANI, CLI, A number). Also contains unique "tag" set by UAC
  • To header identifies recipient of the call, usually contains Called ID (called number, B dtmf 500 server internal error. Also contains unique "tag" set by UAS
  • Via header saves information about proxy servers - server IP address and port - to pass response from UAS back to UAC via same set of proxy servers
  • Record-Route header is optionally inserted by proxy servers to route subsequent SIP requests for the same SIP call
  • Call-ID header identifies SIP call along with "From" and "To" headers
  • Contact header specifies IP address and port to be used by peer SIP endpoints for subsequent SIP requests for the same SIP call
  • CSeq header contains a decimal number which increases by 1 with every subsequent SIP request within the same SIP call, with the exception of CANCEL and ACK requests which use the CSeq number of corresponding INVITE request
  • Server header indicates model, software/firmware version of UAS
  • User-Agent header indicates model, software/firmware version of UAC
  • Content-Length header as you will read later, INVITE and "200 OK" packets usually contain encapsulatedSDP protocol data with information dtmf 500 server internal error RTP audio streams. This header specifies size of encapsulated SDP (or other) data
  • Content-Type header specifies type of encapsulated data. Equals to "application/sdp" in case of encapsulated SDP
  • Allow header lists set of SIP requests supported by user agent which generated the SIP message
  • Supported header specifies set of supported SIP extensions by user agent which generated the SIP message
  • Require header specifies set of required SIP extensions by user agent which generated the SIP message
  • Session-Expires header - part of RFC 4028 (Session Timers) SIP extension - specifies max. interval to refresh SIP call error 017 undefined symbol color_white keep-alive packet (the RFC3261 standard does not define way to abort hang calls)
  • Expires header specifies lifetime of corresponding SIP request
  • WWW-Authenticate, Authorization headers are used for digest authentication
There are many more headers; adding custom headers into requests and responses is a way to extend SIP protocol.
Tasks:
  • See SIP headers in this file: what software is used for UAC and UAS? Dtmf 500 server internal error is SIP Call ID?
  • See how to fix Contact header sent by UAS in this file
Registration
Registration in SIP means passing information about user agent to server, it is implemented with REGISTER. It means "Hello SIP server, here are my contact details for extension number 456: IP address 1.2.3.4, SIP port 5060". Without registration SIP phones must know IP addresses of every destination phone; with registration the IP lync 2010 address book sync error are stored by SIP server, also known as SIP registrar server. Her is example SIP trace for the registration process:
SIP messages for Call-ID 64b10daa51dd4c5a9d6038a43163b22d

14:32:00.599 [+0,00ms] [TX] REGISTER to 5.135.179.50:6000
REGISTER sip:startrinity.com:6000 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.4:5090;rport;branch=z9hG4bKPj609d6772ba754dcab9cba92bcb34be69
Max-Forwards: 70
From: <sip:[email protected]>;tag=aef6297c37914f69832efc2c44c980aa
To: <sip:100@startrinity.com>
Call-ID: 64b10daa51dd4c5a9d6038a43163b22d
CSeq: 22151 REGISTER
Allow: INFO, PRACK, dtmf 500 server internal error, SUBSCRIBE, NOTIFY, REFER, dtmf 500 server internal error, INVITE, ACK, BYE, CANCEL, UPDATE
Contact: <sip:[email protected]:5090>
dtmf 500 server internal error User-Agent: StarTrinity.SIP 2018-06-14 19.29 UTC
Expires: 3600
Content-Length: 0


============================================================end of message=================
14:32:00.674 [+74,84ms] [RX] Unauthorized from 5.135.179.50:6000
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.117.173.127:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPj609d6772ba754dcab9cba92bcb34be69
Call-ID: 64b10daa51dd4c5a9d6038a43163b22d
From: <sip:[email protected]>;tag=aef6297c37914f69832efc2c44c980aa
To: <sip:[email protected]>
CSeq: 22151 REGISTER
WWW-Authenticate: Digest realm="*",nonce="6ab702ae1b742592",opaque="66982f5173f64308",algorithm=md5
Content-Length: 0


============================================================end of message=================
14:32:00.674 [+74,88ms] [TX] REGISTER to 5.135.179.50:6000
REGISTER sip:startrinity.com:6000 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.4:5090;rport;branch=z9hG4bKPjcfb02f366c7f435ba7991cbb13b9497c
Max-Forwards: 70
From: <sip:[email protected]>;tag=aef6297c37914f69832efc2c44c980aa
To: <sip:[email protected]>
Call-ID: 64b10daa51dd4c5a9d6038a43163b22d
CSeq: 22152 REGISTER
Allow: INFO, dtmf 500 server internal error, PRACK, SUBSCRIBE, NOTIFY, dtmf 500 server internal error, REFER, INVITE, ACK, BYE, CANCEL, UPDATE
Contact: <sip:[email protected]:5090>
User-Agent: StarTrinity.SIP 2018-06-14 19.29 UTC
Expires: 3600
Authorization: Digest username="100", realm="*", nonce="6ab702ae1b742592", uri="sip:startrinity.com:6000", response="abd79c866205325d0511a95b1e5f5087", algorithm=md5, opaque="66982f5173f64308"
Content-Length: 0


============================================================end of message=================
14:32:00.758 [+158,50ms] [RX] OK from 5.135.179.50:6000
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.117.173.127:5090;rport=5090;received=87.117.173.127;branch=z9hG4bKPjcfb02f366c7f435ba7991cbb13b9497c
Call-ID: 64b10daa51dd4c5a9d6038a43163b22d
From: <sip:[email protected]>;tag=aef6297c37914f69832efc2c44c980aa
To: <sip:[email protected]>
CSeq: 22152 REGISTER
Contact: sip:[email protected]:5090;expires=3600
initialize post error manager pem Content-Length: 0


============================================================end of message=================
===============================saved by StarTrinity SIP Tester at dtmf 500 server internal error 14:32:12======
As you can see in the SIP trace, first REGISTER request sent by UAC 192.168.10.4 to UAS 5.135.179.50 means resitration of extension 100 at SIP server startrinity.com on SIP port 6000. The UAC sends its IP address in Contact header: Contact: <sip:[email protected]:5090>. "Expires" SIP header equals to 3600 means that the registration entry should be valid for 3600 seconds - in 60 minutes the registrar server should clear the registration entry if not received new REGISTER requests for this extension. REGISTER request with "Expires: 0" means unregistration.
Digest authentication
As you can see in previous SIP trace (with REGISTER), first request is rejected with status "401 Unauthorized" and WWW-Authenticate header, it is called digest challenge. The WWW-Authenticate SIP header contains nonce and opaque values which are used by UAC along with user name and password to send second INVITE request. The second INVITE request contains Authorization header with digest response, it is used by UAS to verify user name and password. If user name and password are correct, UAS sends "200 OK" response, otherwise it sends "403 Unauthorized" response, dtmf 500 server internal error.
SIP extensions
Extensions of SIP protocol are defined by additional RFC standard documents (beyond RFC3261), you can find them here. Most important SIP extensions are
  • 100rel extension defines PRACK method to dtmf 500 server internal error reliable delivery of provisionalSIP responses. It is fully described in RFC3262. error of typical computer solution
    error dbfntx/1001 dos error 4 The PRACK packets are sent in response to 18x provisional responses via chain of SIP proxy servers from UAC to UAS:
    PRACK packets going via SIP proxies
    Here is example SIP call flow for PRACK:
    PRACK call flow
    Without the PRACK, 18x SIP responses could be lost in IP network between UAS and UAC, it means no ringback tone played to caller. With PRACK, 18x SIP response is retransmitted by UAS if it does not receive PRACK from UAC.
  • timer extension defines a keep-alive mechanism for SIP calls, it is described in RFC4068 (Session Timers). The keep-alive packets are sent as re-INVITE or UPDATE SIP packets between UAC and UAS; if the keep-alive packet is not received within session interval, UAC or UAS considers call as hang (dead) and drops the call. Without the session timer extension there could appear hang calls, existing on one SIP UA and already destroyed on another SIP UA. Such situation can happen if internet connection drops between UAC and UAS during end of call, and BYE packet gets lost, dtmf 500 server internal error.

Call transfer

The VX Integrated Media Platform provides the following types of call transfers:

Note: In this topic, the A leg refers to the initial call made to or from the VX Integrated Media Platform.

Bridge transfer

A bridge transfer occurs when the VX Integrated Media Platform initiates a transfer between two parties, and the caller returns to the VX Integrated Media Platform after the transfer ends (when the caller disconnects from the third party). With a bridge transfer:

  • The VX Integrated Media Platform is aware of the outcome of the transfer.

  • The original caller is not disconnected in the event of a connection error.

A bridge transfer appears as a new SIP INVITE from the VX Integrated Media Platform. The audio mixing occurs in the Media Resource Function component of the VoiceXML Interpreter. This component also performs DTMF hotword detection on the A leg.

The dest attribute has different meanings, depending on whether the call is being transferred to dtmf 500 server internal error SIP target or a PSTN target:

For this transfer target.

The dest attribute specifies the.

SIP

Destination SIP URI of the transfer target.

Place this SIP call by using the com.vision.miosip.rvsip.outboundProxyHost configuration setting.

PSTN

User part of the To SIP URI.

The remainder of the URI is constructed using the URI from the com.vision.miosip.dialog.TelHosts configuration setting.

The dest attribute is specified using the tel: URI syntax and can include the custom parameters connecttimeout and maxtime. A tel: URI is converted to a SIP URI, with the user=phone parameter indicating that the call must be placed over the PSTN network.


The following example shows a tel: URI, along with its custom parameters:

tel:12345678;maxtime=60s&connecttimeout=10s.

The following table describes the custom parameters:

Custom parameter

Description

connecttimeout

Amount of time to wait for a final SIP response to the Dtmf 500 server internal error. When this time limit is reached, a CANCEL is issued and the transfer aborted. The timer for the connecttimeout starts when the 180 Ringing response is received.

maxtime

Maximum duration allowed for a call. When this time limit is reached, a BYE is issued to the outbound leg. The SIP INVITE contains the custom header Vision-ParentCallID, which equates to the Call-ID of the A leg.


While the outbound call is being attempted, the A leg hears the audio specified by the com.vision.miosip.defaultTransferAudio configuration setting.

The following table describes the mapping of SIP responses from the INVITE to the value of either the transfer form item variable or the VoiceXML event:

SIP response

Transfer tmp/mysql.sock error 2002 item variable / VoiceXML event

404 Not Found

error.connection.baddestination

408 Request Timeout

noanswer

486 Busy Here

busy

500 Server Internal Error

unknown

503 Service Unavailable

noresource

603 Decline

noanswer


The following table describes how the transfer form item variable is set when the outbound leg of a call is terminated:

Action

Transfer form item variable setting

If the called party hangs up (issues a BYE), the outbound call terminates.

far_end_disconnect

VoiceXML Interpreter terminates the outbound leg because the maximum call duration expired or because a hotword was detected.

near_end_disconnect


Blind transfer

A blind transfer occurs when the VX Integrated Media Platform initiates a transfer between two parties and then detaches from the call before the transfer takes dtmf 500 server internal error. With a blind transfer:

  • The VX Integrated Media Platform is not aware of the outcome of the transfer.

  • The original caller is disconnected if there is a transfer error.

A blind transfer uses the SIP REFER method (RFC 3515). After a successful response to the REFER message, the VX Integrated Media Platform generates a connection.disconnect.transfer and issues a BYE to drop the call.

The following table describes the attributes of the transfer element:

Attribute

Description

dest

Sets the user part of the SIP URI specified by the Refer-To header in the REFER message.

aai

Sets the aai part of the SIP URI specified by the Refer-To header in the REFER message.


The following example shows the use of a REFER message:

REFER sip:[email protected]:5060 SIP/2.0
From: <sip:[email protected]>
To: <sip:10.3.1.52>;tag=ds-42d6-bb2c
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 1 REFER
Content-Length: 0
Refer-To: sip:[email protected];connecttimeout=2s;maxtime=60s
Via: SIP/2.0/UDP 10.0.0.99:5060

Valid responses to the REFER message are:

  • 202 Accepted

  • 500 Server Internal Error if an error occurs

Consultation transfer

A consultation transfer occurs when the VX Integrated Media Platform initiates a transfer between two parties and then detaches from the call after the transfer takes place. With a consultation transfer:

A consultation transfer is also called a supervised transfer.

The following table describes the consultation transfer process:

Task

Description

1

The VX Integrated Dtmf 500 server internal error Platform creates a new call leg by issuing an INVITE message to the outbound server.

2

If the server accepts the call, the VX Integrated Media Platform issues a REFER message on the new call leg, dtmf 500 server internal error.

The Refer-To header contains a Replaces field that holds the Error code 0x800f0826 windows 7 sp1 ID value for the new leg. This dtmf 500 server internal error the new call leg to be replaced by the initial call leg on the remote UAC.

3

Control returns to the VoiceXML context.

4

A <connection.disconnect.transfer> event is issued.



The Phone service is one of the basic functions of the intercom: helps you establish connections with other IP network terminal equipment. The 2N IP intercoms support the extended SIP and are compatible with and certified by the leading SIP PBX and terminal equipment manufacturers (CISCO, Avaya, Broadsoft, etc.).

The intercom supports up to five parallel calls: 1 outgoing and up to 4 incoming calls. Just one of the calls can be active – the audio stream is interconnected with the microphone and speaker and video stream with the camera. The other calls are dtmf 500 server internal error the microphone and speaker are muted, the intercom receives the DTMF characters for the opponent to control the intercom (activate/deactivate profiles, users, etc.).

Typically, the intercoms are used for outgoing calls and incoming calls are inactive – the microphone and speaker are muted. However, you can configure your intercom to make incoming calls active and ringing; refer java.lang.outofmemoryerror bitmap size exceeds vm budget the Calls tab. Press the * and # keys on the numeric keypad to answer and terminate an incoming call. 

The 2N IP intercoms use the G.711L16, G.722 and G.729 protocols to encrypt or compress audio streams and the H.263 or H.264 codecs to compress video streams. Broadband codecs L16 and G.722 are available in selected 2N IP intercom models only. Choose your preferential codecs in the Audio or Video tab.

Explanation of IP Telephony Terms

  • SIP (Session Initiation Protocol) – is a phone call signalling transmission protocol used in IP telephony. It is primarily used for setting up, terminating and forwarding calls between two SIP devices (the intercom and 5a internal cpu error IP phone in this case). SIP devices can establish connections directly with each other (Direct SIP Call) or, typically, via one or more servers: SIP Proxy and SIP Registrar.
  • SIP Dtmf 500 server internal error is an IP network server responsible for call routing (call transfer to another entity closer to the destination). There can be one or more SIP Proxy units between the users.
  • SIP Registrar – is an IP network server responsible for user registration in a certain network section. As a rule, SIP device registration is necessary for a user to be accessible to the others on a certain phone number. SIP Registrar and Error setting display mode createdevice failed d3derr_notavailable Proxy are often installed on one and the same ros brown-driver install error Transport Protocol) – is a protocol defining the standard packet format for audio and video transmission in IP networks. 2N IP intercom uses the RTP for audio and video stream transmission during a call. The stream parameters (port numbers, protocols and codecs) are defined and negotiated via the SDP (Session Description Protocol).

The 2N IP intercoms support three ways of SIP signalling:

  • via the User Datagram Protocol (UDP), which is the most frequently used unsecured signalling method
  • via the Transmission Control Protocol (TCP), which is less frequent, dtmf 500 server internal error, yet recommended unsecured signalling method
  • via the Transaction Layer Security (TLS) protocol, where SIP messages are secured against third party monitoring and modification (except models 2N® IP Base, Uni)

List of Parameters

The 2N IP intercom Phone settings are arranged in five tabs:

  • SIP 1 and SIP 2 – complete SIP terminal settings
  • Calls – incoming and outgoing call settings
  • Audio – audio codec, DTMF transmission and other audio stream transmission settings
  • Video – video codec, video resolution and other video stream transmission settings
  • Local Calls – set the local calls including connections, video parameters 
  • Calling to ACS – set the Axis Dtmf 500 server internal error Station calls

SIP 1 and SIP 2

The 2N IPintercoms allow two independent SIP accounts (SIP 1 and SIP 2 tabs) to be configured, dtmf 500 server internal error. Thus, the intercom can be registered under two phone numbers, with two different SIP exchanges and so on. Both the SIP accounts process incoming calls equivalently. Outgoing calls are primarily processed by account 1, or, if account 1 is not registered (due to SIP exchange error, e.g.), by account 2. Select the account number for the phone numbers included in the phone directory to specify the account to be used for outgoing calls (example: 2568/1 - calls to number 2568 go via account 1, sip:[email protected] calls to sip uri via account 2).


  • Display Name – set the name to be displayed as CLIP on the called party's phone.
  • Phone Number (ID) – set the intercom phone number (or another unique ID including characters and digits). Together with the domain, this number represents a unique intercom identification in calls and registration. 
  • Domain – set the domain name of the service with which the intercom is registered. Typically, it is identical with the SIP Proxy or Registrar address.
  • Test Call – display a dialogue window enabling you to make a test call to a selected phone number, see below. 


  • Use Authentication ID – enable the use of an alternative ID for intercom authentication. If disabled, the phone number defined above is used for authentication.
  • Authentication ID – enter the alternative ID for authentication.
  • Password – enter the password for authentication. The parameter is applied on if your PBX requires authentication.

  • Proxy Address – set the SIP Proxy IP address or domain name.
  • Proxy Port* – set the SIP Proxy port. The device uses the default port according to the transport layer (5060 or 5061) or a port obtained from DNS in case the parameter is empty or set to 0.
  • Backup Proxy Address – set the SIP 442 cisco vpn error windows 7 IP address or domain name to be used where the main proxy dtmf 500 server internal error to respond to requests.
  • Backup Proxy Port* – set the backup SIP Proxy port. The bind socket error uses the default port according to the transport layer (5060 or 5061) or a port obtained from DNS in case the parameter is empty or set to 0.

  • Registration Enabled – enable intercom registration with the set SIP Registrar.
  • Registrar Address – set the SIP Registrar Dtmf 500 server internal error address or domain name.
  • Registrar Port*– set the SIP Registrar port. The device uses the default port according to the transport layer (5060 dtmf 500 server internal error 5061) or a port obtained from DNS in case the parameter is empty or set to 0.
  • Backup Registrar Address – set the SIP registrar IP address or domain name to be used where the main registrar fails to respond to requests.
  • Backup Registrar Port* – set the backup SIP registrar port. The device uses the default port according to the transport layer (5060 or 5061) or a port obtained from DNS in case the parameter is empty or set to 0.
  • Registration Expires – define the registration expiry, which affects the network and SIP Registrar load by periodically sent registration requirements. The SIP Registrar can modify the expiry limit without letting you know.
  • Registration State – display the current registration state (unregistered, registering., registered, unregistering.).
  • Failure Reason – display the contr terrorism 3d episode for the last registration attempt failure: the last error reply of the registrar, e.g. 404 Not Found.
     

Tip

  •  To set the Outbound Proxy complete the Outbound Proxy address into the Proxy address and Registrar address parameters. Domain = Registrar address.

Caution

  • If the parameter* is empty or set to 0, the default port is used according to the selected transport protocol (5060 for TCP or UDP, 5061 for TLS).

  • SIP Transport Protocol – set the SIP communication protocol: UDP (default), dtmf 500 server internal error, TCP or TLS.
  • Lowest Allowed TLS Version – define the lowest TLS version to be connected to the devices.
  • Verify Server Certificate – verify the SIP server public certificate against the CA certificates uploaded in the device.
  • Client Certificate  specify the client certificate and private key used for verifying the intercom’s authority to communicate with the SIP server.
  • Local SIP Port – set the local port to be used for SIP signalling. The parameter is not applied until the intercom is restarted. The default value is 5060.
  • PRACK Enabled – enable the PRACK method for reliable confirmation of SIP messages with codes 101–199.
  • REFER Enabled – enable call forwarding via the REFER method.
  • Send KeepAlive Packets – define whether the intercom shall, during a call, send periodical SIP OPTIONS requests to inquire about the state of the called station (to detect the station failure, e.g.).
  • IP Address Filter Enabled – enable the blocking of SIP packet receiving from addresses other than SIP Proxy and SIP Registrar. The primary purpose of the function is to enhance communication security and eliminate unauthorised phone calls.
  • Receive Encrypted Calls Only (SRTP) – set that SRTP encrypted calls shall only be received on this account. Unencrypted calls will be rejected. At the same time, TLS is recommended as the SIP transport protocol for higher security.
  • Encrypted Outgoing Calls (SRTP) – set that outgoing calls shall be SRTP encrypted on this account. At the same time, TLS is recommended as the SIP transport protocol for higher security.
  • Do Not Play Incoming Early Media – disable playing of the incoming audio stream before call pick-up (early media), which is sent by some PBXs or other devices. A standard local ringtone is played instead.
  • QoS DSCP Value – set the SIP packet priority in the network. The set value is sent in the TOS (Type of Service) field in the IP packet header. Value is entered in decimal format. The parameter is not applied until the intercom is restarted.
  • Starting RTP Port – set the starting local RTP port in the range of the length of 64 ports to be used for audio and video transmissions. The default value is 4900 (i.e. the used range is 4900–4963). The parameter is only set for account 1 but applies to both the SIP accounts.
  • External IP Address – set the public IP address or name of the router to which your intercom is connected. If the intercom IP address is public, leave this field blank.
  • RTP Timeout – set the audio stream RTP packet receiving timeout during a call, dtmf 500 server internal error. If this limit is exceeded (RTP packets are not delivered), the call is terminated by the intercom. Set the parameter to 0 to disable this function. The parameter is only set for account 1 but applies to both the SIP accounts.
  • Compatibility with Broadsoft devices – set the Broadsoft PBX compatibility mode. Having received re-invite from a Dtmf 500 server internal error in this mode, the intercom replies by repeating the last sent SDP with currently used codecs instead of sending a complete offer.
  • Rotate SRV record – Allow SRV record rotation for SIP Proxy and Registrar. This is an alternative method of transition to backup servers in the event of main server failure or unavailability.

Caution

  • To use the NAPTR / SRV DNS query, cancel the Proxy/Registrar port setting.

Calls


  • Call Time Limit – set the call time limit after which the call is automatically terminated. The intercom signals termination with a 10s beep before the call end. Enter any DTMF character into the call (# on your IP phone, e.g.) to extend the call time. If the call duration is set to 0 and SRTP is not used, the call is not time limited.

  • Call Answering Mode(SIP1, SIP2) – set the incoming call receiving mode. The following three options are available:
    • Always busy – the intercom rejects incoming calls,
    • Manual – the intercom alerts incoming calls and the user answers them using a numeric keypad button, and 
    • Automatic – the intercom answers incoming calls automatically. You can set the call receiving mode for each SIP account separately.
    • Automatic (DTMF only) – the intercom answers incoming calls automatically only if DTMF without connection to a microphone and speaker is received.
  • Local Call Receiving Mode – set the incoming local call receiving mode
    • Always busy – the intercom rejects incoming calls,
    • Manual – the intercom alerts incoming calls and the user answers them using a numeric keypad button, and 
    • Automatic – the intercom answers incoming calls automatically. You can set the call receiving mode for each SIP account separately.
    • Automatic (DTMF only) – the intercom answers incoming calls automatically only if DTMF without connection to a microphone and speaker is received.
  • Pick Up in – set the timeout after which the call is automatically picked up in the automatic call answering mode. If one of the Answering machine modes is enabled in an Answering machine supporting device, the call is picked up after the timeout and the selected voice message is played in both the automatic and manual call answering modes. If this value is 0, the voice message is played instantaneously, dtmf 500 server internal error. Shared by all the SIP accounts.
  • Answer Incoming Call by Button – pick up an incoming call via a selected speed dial button. Set to None to disable the function.

Caution

  • The Answer Incoming Call by Button function is not displayed in the keypad-equipped 2N® IP Force and 2N® IP Vario models. With these models, answer incoming calls by pressing the green-earpiece button on the keypad without prior configuration.

  • Ring Time Limit – set the outgoing call setup and ringing time limit after which the calls shall be automatically terminated. If the calls are routed to the GSM network via GSM gateways, you are advised to set a value higher than 20 s. Minimum value 1 s, maximum value 600 s. Configure 0 to disable this time limit.
  • Dial Cycles Limit – set the maximum count of user deputy dial cycles if the user dialled by the Phone Book position number is inaccessible. The function helps you avoid deadlock if the User Deputy is set to the same value in the Phone Book. Refer to Subs. 5.4.1.1 Calling Cycle Limit for calling cycle limit settins options.
  • Calling Virtual Numbers – allow the calling of preset virtual numbers of users.
  • Floor/Apartment Dialing Mode – enable the special Floor/Apartment dialling mode. In this mode, enter the assigned user virtual number via the numeric keypad.  Available for model 2N® IP Vario only. Enter the floor/apartment code to the user Virtual number. The code may include digits and letters A–F.
  • Telephone Mode Enabled  enable the option to set up calls directly to the phone numbers dialed via the intercom numeric keypad. Enter the phone number key sequence to set up the call.

  • Maximum Number of Dialed Digits  Set the maximum avira error 559 of digits for a phone number in the Telephone mode. When this limit is reached, the number is dialed automatically without pressing *.
  • Button Function During Outgoing Call – set the quick dial button function during an outgoing call. You can only set the button that initiated the call.

  • Enable Crestron Network Discovery – enable 2N IP intercom identification within the Crestron network.
  • Crestron Device Name – select the device name.
  • Crestron Group List – select the group name list with comma as a separator.
  • Enable Video Multicast for Crestron panels – enable video multicast for Crestron panels, allowing for multiple Crestron devices to receive the same video stream without wasting the local network 0x00000be error windows 7 Multicast Address – set the multicast address to be used for multicast video for Crestron devices.
  • Crestron Multicast Port – set the multicast port to be used for multicast video for Crestron devices.
  • Crestron Multicast TTL – set the Time To Live (TTL) value to be used for sending video early media for Crestron devices. 

Audio


  • Enable/disable the use of audio codecs for call setups and set their priorities. Broadband codecs L16 and G.722 are available in selected intercom models only. Codec G.729 is available for all the 2N IP intercoms.

The tab below helps you define how DTMF characters shall be sent from the intercom. Check the DTMF dtmf 500 server internal error options and settings of the opponent to make the function work properly.

  • Sending Mode – define whether it will be possible to send DTMF during a call by pressing 0 through 9, * and # on the intercom numeric keypad. Set the sending mode for incoming/outgoing/all calls.
  • In-Band (Audio) – enable classic DTMF error bptm overland no robot daemon tone sending in the audio band.
  • RTP (RFC-2833) – enable DTMF sending via the RTP according to RFC-2833.
  • SIP INFO (RFC-2976) – enable DTMF sending via SIP INFO messages according to RFC-2976.

The tab below helps you define how DTMF characters shall be received from the intercom. Check the DTMF receiving options and settings of the opponent to make the function work properly.

  • In-Band (Audio) – enable classic DTMF dual tone receiving in the audio band.
  • RTP (RFC-2833) – enable DTMF receiving via the RTP according to RFC-2833.
  • SIP INFO (RFC-2976) – enable DTMF receiving via SIP INFO messages according to RFC-2976.

  • QoS DSCP Value – set the audio RTP packet priority in the network. The set value is sent in the TOS (Type of Service) field in the IP packet header. Value is entered in decimal format. The parameter is not applied until the intercom is restarted.
  • Jitter Compensation – set the buffer capacity for jitter compensation in audio packet transmissions. A higher capacity improves the transmission resistance at the dtmf 500 server internal error of a greater sound delay.

Video


  • Enable/disable the use of video codecs for call setups and set their priorities.

  • Video Resolution – set the video resolution for phone calls.
  • Video Framerate – set the video frame rate for phone calls.
  • Video Bitrate – set the video stream bit rate for phone calls.

  • PTZ Mode – enable the PTZ (Pan-Tilt-Zoom) function to control the camera display area during the call via DTMF (Enhanced Video license required) from your IP phone numeric keypad. Click the * key to enable/disable the PTZ mode. The meanings of the IP phone keys in the PTZ mode are as follows:

    *Enable/disable PTZ
    1Zoom in
    3Zoom out
    2Move zoom region up
    4Move zoom region to the left
    6Move zoom region to the right
    8Move zoom region down
    5Return to initial state

  • QoS DSCP Value – set the video RTP packet priority in the network. The set value is sent in the TOS (Type of Service) field in the IP packet header.
  • Maximum Packet Size – set the size limit for the video RTP packets to be sent.

  • H.264 Payload Type (1) – set the payload type for video codec H.264 (packetisation mode 1). Set a value from the range of 96 through 127, or 0 to disable this codec type.
  • H.264 Payload Type (2) – set the payload type for video codec H.264 (packetisation mode 2). Set a value from the range of 96 through 127, or 0 to disable this codec type.
  • H.263+ Payload Type – set the payload type for video codec H.263+ (packetisation mode 3). Set a value from the range of 96 through 127.
  • Use the sendrecv attribute for video – the setting was earlier named Compatibility with Polycom phones. This setting provides compatibility with some third party devices (Polycom/Cisco and others). In this mode, the intercom sends sendrecv instead of sendonly in the SDP message in the codec offer for video.

Tip

  • For the Video Preview feature at the Grandstream GXV 3275 phone (video transferred via Early Media) no configuration is needed. Check your PBX vendor whether this feature is supported by your PBX system.
  • For the Video Preview feature at the Gigaset Maxwell 10 phone (video transferred via jpg images) it is necessary to set Connection Type to Unsecure and Authentication to None at the Camera API in HTTP API.

Local Calls

This tab contains settings for connection of the 2N answering units to the intercom. The main parameter is the access key, which secures the connection and enables you to create multiple independent groups of intercoms and 2N answering units within the local network. It also contains the video transmission settings.

  • Enable Local Calls – enable calls between 2N devices in the LAN. With this function off, the other LAN devices cannot locate this device, i.e. cannot call the device in the device:device_ID dtmf 500 server internal error width="500" src="https://wiki.2n.com/hip/conf/files/latest/en/74744952/74745072/1/1656330090372/image2019-3-28+10%3A50%3A43.png">

  • Device ID – set the device ID to be displayed in the LAN device list in all the 2N devices in one and the same LAN, dtmf 500 server internal error. You can direct a call to this device by setting the user phone number as device:device_ID in these devices.

  • Access Key 1–3 – set the access key to be shared by the intercom and 2N answering unit. If the access keys do not match in the intercom and2N answering unit, the intercom cannot call the 2N answering unit and the 2N answering unit cannot receive video from the intercom. Each intercom can be assigned up to three access keys and thus become a member of up to three independent 2N answering unit groups. The Access key length is up to 63 characters. 
  • The access key cannot be used with 2N® Indoor Touch firmware v. 2 or 3 where it has to be set as empty. The access key can only be used for 2N® Indoor Touch version 4 or higher.

 


  • Video Resolution – set the resolution of the video dtmf 500 server internal error to be sent to 2N answering unit.
  • Video Framerate – set the framerate of the video stream to be sent to 2N answering unit.
  • Video Quality – set the quality of the MJPEG video stream to be sent to 2N answering units.
  • Multicast Group – set the multicast address to which the intercom video stream shall be sent. Select one of the 8 preset addresses or set the mode in which the intercom selects the address automatically. 
  • Enable Video Preview – enable video preview multicast transmission.

  • LAN Device Count– display the current count of local 2N answering units connected to the intercom, i.e. those registered with the intercom.
  • Number of Listening/Watching Devices – display the current count of 2N answering units watching video streams from the intercom.
  • Show LAN device list – display the list of local 2N answering units.

ACS

This function is used for integration of the Axis Camera Station service into the 2N IP intercoms.


  • Enable ACS Call – allow calling to the Axis Camera Station (ACS). Use a special URI in the vms:* format for the ACS calls.

Caution

  • In case dtmf 500 server internal error 2N IP intercom has already been added to the ACS, back up all of its records before upgrade, then remove the 2N IP intercom from the ACS, perform upgrade and add the intercom again.

  • Username – ACS call authentication username.
  • Password – ACS call authentication password.

Asterisk Community

Hi Mr,
when i make cammand, i recived the result.

I see : From: “Anonymous” sip:[email protected];tag=as21a80c44
What should I do? if i want the result:
From: “199” sip:[email protected];tag=as21a80c44

localhostCLI>
localhost
CLI>
localhost*CLI> originate SIP/[email protected]:5060 extension [email protected]
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.226.2.10:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2cd850e0
Max-Forwards: 70
From: “Anonymous” sip:[email protected];tag=as21a80c44
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.5.0
Date: Thu, 26 Aug 2021 07:34:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1365028515 1365028515 IN IP4 10.226.39.49
s=Asterisk PBX 1.8.5.0
c=IN IP4 10.226.39.49
t=0 0
m=audio 48800 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


<— SIP read dtmf 500 server internal error UDP:10.226.2.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2cd850e0
Call-ID: [email protected]:5060
From: "Anonymous"sip:[email protected];tag=as21a80c44
To: sip:[email protected]:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:10.226.2.10:5060 —>
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2cd850e0
Call-ID: [email protected]:5060
From: "Anonymous"sip:[email protected];tag=as21a80c44
To: sip:[email protected]:5060;tag=z66141d4
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
– Got SIP response 500 “Server Internal Error” back from 10.226.2.10:5060
Transmitting (no NAT) to 10.226.2.10:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2cd850e0
Max-Forwards: 70
From: “Anonymous” sip:[email protected];tag=as21a80c44
To: sip:[email protected]:5060;tag=z66141d4
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.5.0
Content-Length: 0


<— SIP read from UDP:10.226.2.2:5066 —>
OPTIONS sip:10.226.39.49:5060 SIP/2.0
Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKtrsnro1ra3okt1ong3gqnu1ss;X-DispMsg=1401
Call-ID: [email protected]
From: sip:10.226.2.2:5060;tag=tn1da3ok-CC-1003-OFC-11
To: sip:10.226.39.49:5060
CSeq: 1 OPTIONS
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 0