Sip error 503

sip error 503

If no trusted peers are configured all connections are accepted. The SmartNode responds with a SIP " Service Unavailable” to requests received from hosts. Whereas IP address configured on the SME to HQ SIP Trunk is pointing towards HQ Publisher which is not a primary call processing server, whereas. When logging in to VS Connect, you may experience an error stating either "SIP " or "SIP " followed by "Service Unavailable." A few different reasons. sip error 503

Sip error 503 - this excellent

Outbound calls error with "all circuits busy" or "congestion":

This is the default configuration of Asterisk regardless of the actual error generated (which is infuriating when you are trying to diagnose the real problem) unless PBX is updated to send back the real error rather than the changed error. This error most commonly occurs when the call is not authenticating properly, at which point check the above in the SIP trunk configuration (If Asterisk, swap username= for defaultuser= to see if this solves the issue. Just because a trunk is showing as registered does not mean it will authenticate correctly.

Outbound calls fail with SIP error (Not Accepted Here) or I-SUP errors 58 (bearer capability not available) or 88 (incompatible destination):

Check the codecs allowed in the SIP trunk configuration above, VoiceHost only supports: alaw, ulaw, gsm
If a codec is defined in Asterisk that is not one of the above, or is offering a differing sample rate or interval rate (e.g. /20i - Hz at 20ms) cannot interwork with /30i - Hz at 30ms) the call will fail and the codecs in the SIP trunk configuration need to be aligned to use one of the above codecs.

Inbound calls fail with SIP error (Request Timeout):

Check the inbound number is mapped in the system correctly, if necessary the SIP trunk on the portal can be configured to strip the plus, e.g. if Asterisk is configured to use plus somewhere else. Check the trunk is registered. Ascertain how long the error took to come back if it was immediate the trunk is usually unregistered if it took a few seconds the number is usually not mapped correctly.

Calls fail with SIP error , I-SUP errors 34 or

If our platform replies back with it usually means the gateway trying to process the call can't due to "issues", or the customer has hit their Calls-Per-Second (CPS) limit and is sending too many calls at once. Sometimes the error is passed back from IP Exchange through VoiceHost to the customer's system, at which point the call will usually hunt to another route to try and place the call.

Cause code (ISUP)SIP EquivalentDefinition
1 Not FoundUnallocated (unassigned) number
2 Not foundno route to network
3 Not foundno route to destination
16BYE or CANCEL (*)normal call clearing
17 Busy hereuser busy
18 Request Timeoutno user responding
19 Temporarily unavailableno answer from the user
20 Temporarily unavailablesubscriber absent
21 Forbidden (+)call rejected
22 Gonenumber changed (w/o diagnostic)
22 Moved Permanentlynumber changed (w/ diagnostic)
23 Goneredirection to new destination
26 Not Found (=)non-selected user clearing
27 Bad Gatewaydestination out of order
28 Address incompleteaddress incomplete
29 Not implementedfacility rejected
31 Temporarily unavailablenormal unspecified
34 Service unavailableno circuit available
38 Service unavailablenetwork out of order
41 Service unavailabletemporary failure
42 Service unavailableswitching equipment congestion
47 Service unavailableresource unavailable
55 Forbiddenincoming calls barred within CUG
57 Forbiddenbearer capability not authorized
58 Service unavailablebearer capability not presently
65 Not Acceptable Herebearer capability not implemented
70 Not Acceptable Hereonly restricted digital avail
79 Not implementedservice or option not implemented
87 Forbiddenuser not member of CUG
88 Service unavailableincompatible destination
Gateway timeoutrecovery of timer expiry
Server internal errorprotocol error
Server internal errorinterworking unspecified

List of SIP response codes

The Session Initiation Protocol (SIP) is a signallingprotocol used for controlling communication sessions such as Voice over IPtelephone calls. SIP is based on request/response transactions, in a similar manner to the Hypertext Transfer Protocol (HTTP). Each transaction consists of a SIP request (which will be one of several request methods), and at least one response.[1]:&#;p11&#;

SIP requests and responses may be generated by any SIP user agent; user agents are divided into clients (UACs), which initiate requests, and servers (UASes), which respond to them.[1]:&#;§8&#; A single user agent may act as both UAC and UAS for different transactions:[1]:&#;p26&#; for example, a SIP phone is a user agent that will be a UAC when making a call, and a UAS when receiving one. Additionally, some devices will act as both UAC and UAS for a single transaction; these are called Back-to-Back User Agents (B2BUAs).[1]:&#;p20&#;

SIP responses specify a three-digit integer response code, which is one of a number of defined codes that detail the status of the request. These codes are grouped according to their first digit as "provisional", "success", "redirection", "client error", "server error" or "global failure" codes, corresponding to a first digit of 1–6; these are expressed as, for example, "1xx" for provisional responses with a code of –[1]:&#;§&#; The SIP response codes are consistent with the HTTP response codes, although not all HTTP response codes are valid in SIP.[1]:&#;§21&#;

SIP responses also specify a "reason phrase", and a default reason phrase is defined with each response code.[1]:&#;§&#; These reason phrases can be varied, however, such as to provide additional information[1]:&#;§&#; or to provide the text in a different language.[1]:&#;§&#;

The SIP response codes and corresponding reason phrases were initially defined in RFC [1] That RFC also defines a SIP Parameters Internet Assigned Numbers Authority (IANA) registry to allow other RFC to provide more response codes.[1]:&#;§27&#;[2]

This list includes all the SIP response codes defined in IETFRFCs and registered in the SIP Parameters IANA registry as of 14&#;July&#;[update]. This list also includes SIP response codes defined in obsolete SIP RFCs (specifically, RFC ), which are therefore not registered with the IANA; these are explicitly noted as such.

1xx—Provisional Responses[edit]

Trying
Extended search being performed may take a significant time so a forking proxy must send a Trying response.[1]:&#;§&#;
Ringing
Destination user agent received INVITE, and is alerting user of call.[1]:&#;§&#;
Call is Being Forwarded
Servers can optionally send this response to indicate a call is being forwarded.[1]:&#;§&#;
Queued
Indicates that the destination was temporarily unavailable, so the server has queued the call until the destination is available. A server may send multiple responses to update progress of the queue.[1]:&#;§&#;
Session Progress
This response may be used to send extra information for a call which is still being set up.[1]:&#;§&#;
Early Dialog Terminated
Can be used by User Agent Server to indicate to upstream SIP entities (including the User Agent Client (UAC)) that an early dialog has been terminated.[3]

2xx—Successful Responses[edit]

OK
Indicates that the request was successful.[1]:&#;§&#;
Accepted
Indicates that the request has been accepted for processing, but the processing has not been completed.[4]:&#;§&#;[5] Deprecated.[6]:&#;§&#;[2]
No Notification
Indicates the request was successful, but the corresponding response will not be received.[7]

3xx—Redirection Responses[edit]

Multiple Choices
The address resolved to one of several options for the user or client to choose between, which are listed in the message body or the message's Contact fields.[1]:&#;§&#;
Moved Permanently
The original Request-URI is no longer valid, the new address is given in the Contact header field, and the client should update any records of the original Request-URI with the new value.[1]:&#;§&#;
Moved Temporarily
The client should try at the address in the Contact field. If an Expires field is present, the client may cache the result for that period of time.[1]:&#;§&#;
Use Proxy
The Contact field details a proxy that must be used to access the requested destination.[1]:&#;§&#;
Alternative Service
The call failed, but alternatives are detailed in the message body.[1]:&#;§&#;

4xx—Client Failure Responses[edit]

Bad Request
The request could not be understood due to malformed syntax.[1]:&#;§&#;
Unauthorized
The request requires user authentication. This response is issued by UASs and registrars.[1]:&#;§&#;
Payment Required
Reserved for future use.[1]:&#;§&#;
Forbidden
The server understood the request, but is refusing to fulfill it.[1]:&#;§&#; Sometimes (but not always) this means the call has been rejected by the receiver.
Not Found
The server has definitive information that the user does not exist at the domain specified in the Request-URI. This status is also returned if the domain in the Request-URI does not match any of the domains handled by the recipient of the request.[1]:&#;§&#;
Method Not Allowed
The method specified in the Request-Line is understood, but not allowed for the address identified by the Request-URI.[1]:&#;§&#;
Not Acceptable
The resource identified by the request is only capable of generating response entities that have content characteristics but not acceptable according to the Accept header field sent in the request.[1]:&#;§&#;
Proxy Authentication Required
The request requires user authentication. This response is issued by proxies.[1]:&#;§&#;
Request Timeout
Couldn't find the user in time. The server could not produce a response within a suitable amount of time, for example, if it could not determine the location of the user in time. The client MAY repeat the request without modifications at any later time.[1]:&#;§&#;
Conflict
User already registered.[8]:&#;§&#; Deprecated by omission from later RFCs[1] and by non-registration with the IANA.[2]
Gone
The user existed once, but is not available here any more.[1]:&#;§&#;
Length Required
The server will not accept the request without a valid Content-Length.[8]:&#;§&#; Deprecated by omission from later RFCs[1] and by non-registration with the IANA.[2]
Conditional Request Failed
The given precondition has not been met.[9]
Request Entity Too Large
Request body too large.[1]:&#;§&#;
Request-URI Too Long
The server is refusing to service the request because the Request-URI is longer than the server is willing to interpret.[1]:&#;§&#;
Unsupported Media Type
Request body in a format not supported.[1]:&#;§&#;
Unsupported URI Scheme
Request-URI is unknown to the server.[1]:&#;§&#;
Unknown Resource-Priority
There was a resource-priority option tag, but no Resource-Priority header.[10]
Bad Extension
Bad SIP Protocol Extension used, not understood by the server.[1]:&#;§&#;
Extension Required
The server needs a specific extension not listed in the Supported header.[1]:&#;§&#;
Session Interval Too Small
The received request contains a Session-Expires header field with a duration below the minimum timer.[11]
Interval Too Brief
Expiration time of the resource is too short.[1]:&#;§&#;
Bad Location Information
The request's location content was malformed or otherwise unsatisfactory.[12]
Bad Alert Message
The server rejected a non-interactive emergency call, indicating that the request was malformed enough that no reasonable emergency response to the alert can be determined.[13]
Use Identity Header
The server policy requires an Identity header, and one has not been provided.[14]:&#;p11&#;
Provide Referrer Identity
The server did not receive a valid Referred-By token on the request.[15]
Flow Failed
A specific flow to a user agent has failed, although other flows may succeed. This response is intended for use between proxy devices, and should not be seen by an endpoint (and if it is seen by one, should be treated as a Bad Request response).[16]:&#;§&#;
Anonymity Disallowed
The request has been rejected because it was anonymous.[17]
Bad Identity-Info
The request has an Identity-Info header, and the URI scheme in that header cannot be dereferenced.[14]:&#;p11&#;
Unsupported Certificate
The server was unable to validate a certificate for the domain that signed the request.[14]:&#;p11&#;
Invalid Identity Header
The server obtained a valid certificate that the request claimed was used to sign the request, but was unable to verify that signature.[14]:&#;p12&#;
First Hop Lacks Outbound Support
The first outbound proxy the user is attempting to register through does not support the "outbound" feature of RFC , although the registrar does.[16]:&#;§&#;
Max-Breadth Exceeded
If a SIP proxy determines a response context has insufficient Incoming Max-Breadth to carry out a desired parallel fork, and the proxy is unwilling/unable to compensate by forking serially or sending a redirect, that proxy MUST return a response. A client receiving a response can infer that its request did not reach all possible destinations.[18]
Bad Info Package
If a SIP UA receives an INFO request associated with an Info Package that the UA has not indicated willingness to receive, the UA MUST send a response, which contains a Recv-Info header field with Info Packages for which the UA is willing to receive INFO requests.[19]
Consent Needed
The source of the request did not have the permission of the recipient to make such a request.[20]
Temporarily Unavailable
Callee currently unavailable.[1]:&#;§&#;
Call/Transaction Does Not Exist
Server received a request that does not match any dialog or transaction.[1]:&#;§&#;
Loop Detected
Server has detected a loop.[1]:&#;§&#;
Too Many Hops
Max-Forwards header has reached the value '0'.[1]:&#;§&#;
Address Incomplete
Request-URI incomplete.[1]:&#;§&#;
Ambiguous
Request-URI is ambiguous.[1]:&#;§&#;
Busy Here
Callee is busy.[1]:&#;§&#;
Request Terminated
Request has terminated by bye or cancel.[1]:&#;§&#;
Not Acceptable Here
Some aspect of the session description or the Request-URI is not acceptable.[1]:&#;§&#;
Bad Event
The server did not understand an event package specified in an Event header field.[4]:&#;§&#;[6]:&#;§&#;
Request Pending
Server has some pending request from the same dialog.[1]:&#;§&#;
Undecipherable
Request contains an encrypted MIME body, which recipient can not decrypt.[1]:&#;§&#;
Security Agreement Required
The server has received a request that requires a negotiated security mechanism, and the response contains a list of suitable security mechanisms for the requester to choose between,[21]:&#;§§–&#; or a digest authentication challenge.[21]:&#;§&#;

5xx—Server Failure Responses[edit]

Internal Server Error
The server could not fulfill the request due to some unexpected condition.[1]:&#;§&#;
Not Implemented
The server does not have the ability to fulfill the request, such as because it does not recognize the request method. (Compare with Method Not Allowed, where the server recognizes the method but does not allow or support it.)[1]:&#;§&#;
Bad Gateway
The server is acting as a gateway or proxy, and received an invalid response from a downstream server while attempting to fulfill the request.[1]:&#;§&#;
Service Unavailable
The server is undergoing maintenance or is temporarily overloaded and so cannot process the request. A "Retry-After" header field may specify when the client may reattempt its request.[1]:&#;§&#;
Server Time-out
The server attempted to access another server in attempting to process the request, and did not receive a prompt response.[1]:&#;§&#;
Version Not Supported
The SIP protocol version in the request is not supported by the server.[1]:&#;§&#;
Message Too Large
The request message length is longer than the server can process.[1]:&#;§&#;
Push Notification Service Not Supported
The server does not support the push notification service identified in a 'pn-provider' SIP URI parameter[22]:&#;§&#;
Precondition Failure
The server is unable or unwilling to meet some constraints specified in the offer.[23]

6xx—Global Failure Responses[edit]

Busy Everywhere
All possible destinations are busy. Unlike the response, this response indicates the destination knows there are no alternative destinations (such as a voicemail server) able to accept the call.[1]:&#;§&#;
Decline
The destination does not wish to participate in the call, or cannot do so, and additionally the destination knows there are no alternative destinations (such as a voicemail server) willing to accept the call.[1]:&#;§&#; The response may indicate a better time to call in the Retry-After header field.
Does Not Exist Anywhere
The server has authoritative information that the requested user does not exist anywhere.[1]:&#;§&#;
Not Acceptable
The user's agent was contacted successfully but some aspects of the session description such as the requested media, bandwidth, or addressing style were not acceptable.[1]:&#;§&#;
Unwanted
The called party did not want this call from the calling party. Future attempts from the calling party are likely to be similarly rejected.[24]
Rejected
An intermediary machine or process rejected the call attempt.[25] This contrasts with the (Unwanted) SIP response code in which a human, the called party, rejected the call. The intermediary rejecting the call should include a Call-Info header with "purpose" value "jwscard", with the jCard[26] with contact details. The calling party can use this jCard if they want to dispute the rejection.

References[edit]

  1. ^ abcdefghijklmnopqrstuvwxyzaaabacadaeafagahaiajakalamanaoapaqarasatauavawaxayazbabbbcbdbebfbgbhbibjbkRosenberg, Jonathan; Schulzrinne, Henning; Camarillo, Gonzalo; Johnston, Alan; Peterson, Jon; Sparks, Robert; Handley, Mark; Schooler, Eve (June ). SIP: Session Initiation Protocol. IETF. doi/RFC RFC
  2. ^ abcdRoach, Adam; Jennings, Cullen; Peterson, Jon; Barnes, Mary (17 April ) [Created January ]. "Response Codes". Session Initiation Protocol (SIP) Parameters. IANA.
  3. ^Holmberg, Christer (May ). Session Initiation Protocol (SIP) Response Code for Indication of Terminated Dialog. IETF. p.&#;1.&#;Abstract. doi/RFC RFC
  4. ^ abRoach, Adam B. (June ). Session Initiation Protocol (SIP)-Specific Event Notification. IETF. doi/RFC RFC
  5. ^Fielding, Roy T.; Gettys, James; Mogul, Jeffrey C.; Nielsen, Henrik Frystyk; Masinter, Larry; Leach, Paul; Berners-Lee, Tim (June ). " Accepted". Hypertext Transfer Protocol -- HTTP/. IETF. sec.&#; doi/RFC RFC
  6. ^ abRoach, Adam (July ). SIP-Specific Event Notification. IETF. doi/RFC RFC
  7. ^Niemi, Aki (May ). " (No Notification) Response Code". In Willis, Dean (ed.). An Extension to Session Initiation Protocol (SIP) Events for Conditional Event Notification. IETF. sec.&#; doi/RFC RFC
  8. ^ abHandley, Mark; Schulzrinne, Henning; Schooler, Eve; Rosenberg, Jonathan (March ). SIP: Session Initiation Protocol. IETF. doi/RFC RFC
  9. ^Niemi, Aki, ed. (). "" Conditional Requset Failed" Response Code". Session Initiation Protocol (SIP) Extension for Event State Publication. IETF. sec.&#; doi/RFC RFC
  10. ^Schulzrinne, Henning; Polk, James (February ). "No Known Namespace or Priority Value". Communications Resource Priority for the Session Initiation Protocol (SIP). IETF. sec.&#; doi/RFC RFC
  11. ^Donovan, Steve; Rosenberg, Jonathan (April ). " Response Code Definition". Session Timers in the Session Initiation Protocol (SIP). IETF. sec.&#;6. doi/RFC RFC
  12. ^Polk, James; Rosen, Brian; Peterson, Jon (December ). " (Bad Location Information) Response Code". Location Conveyance for the Session Initiation Protocol. IETF. sec.&#; doi/RFC RFC
  13. ^Rosen, Brian; Schulzrinne, Henning; Tschofenig, Hannes; Gellens, Randall (September ). " (Bad Alert Message) Response Code". Non-interactive Emergency Calls. IETF. sec.&#; doi/RFC RFC
  14. ^ abcdPeterson, Jon; Jennings, Cullen (August ). Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP). IETF. doi/RFC RFC
  15. ^Sparks, Robert J. (September ). "The Provide Referrer Identity Error Response". The Session Initiation Protocol (SIP) Referred-By Mechanism. IETF. sec.&#;5. doi/RFC RFC
  16. ^ abJennings, Cullen; Mahy, Rohan; Audet, Francois, eds. (October ). Managing Client-Initiated Connections in the Session Initiation Protocol (SIP). IETF. doi/RFC RFC
  17. ^Rosenberg, Jonathan (December ). " (Anonymity Disallowed) Definition". Rejecting Anonymous Requests in the Session Initiation Protocol (SIP). IETF. sec.&#;5. doi/RFC RFC
  18. ^Addressing an Amplification Vulnerability in Session Initiation Protocol (SIP) Forking Proxies. IETF. December doi/RFC RFC
  19. ^Session Initiation Protocol (SIP) INFO Method and Package Framework. IETF. January doi/RFC RFC
  20. ^Rosenberg, Jonathan; Willis, Dean (October ). "Definition of the Response Code". In Camarillo, Gonzalo (ed.). A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP). IETF. sec.&#; doi/RFC RFC
  21. ^ abArkko, Jari; Torvinen, Vesa; Camarillo, Gonzalo; Niemi, Aki; Haukka, Tao (January ). Security Mechanism Agreement for the Session Initiation Protocol (SIP). IETF. doi/RFC RFC
  22. ^Push Notification with the Session Initiation Protocol (SIP). IETF. May doi/RFC RFC
  23. ^Rosenberg, Jonathan (October ). "Refusing an offer". In Camarillo, Gonzalo; Marshall, Bill (eds.). Integration of Resource Management and Session Initiation Protocol (SIP). IETF. sec.&#;8. doi/RFC RFC
  24. ^A SIP Response Code for Unwanted Calls. IETF. July doi/RFC RFC
  25. ^A Session Initiation Protocol (SIP) Response Code for Rejected Calls. IETF. December doi/RFC RFC
  26. ^RFC&#;

External links[edit]

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SIP Error Response = Service Unavailable

SIP Error Response = Service Unavailable

Bonker(TechnicalUser)

(OP)

Hi All,
I have a issue, I have an IP Office connected via SIP to Session Manager. When I make a call from CM to the IP Office i get the below error Response = Service Unavailable.

The the trunks on the IP Office is configure as SMLine.


2/13/15 PMms Line = 9, Channel = 1, SIP Message = Response, Direction = From Switch, From = [email protected], To = @sprers.eu, Response = Service Unavailable
2/13/15 PMms Line = 9, Channel = 1, SIP Message = Ack, Direction = To Switch, From = [email protected], To = @sprers.eu

Red Flag Submitted

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Knowledgebase

One of the main reasons for this is that the SmartNode does not allow for any "untrusted" host IP Addresses to pass through whenever the "trust" command is configured in the SIP interface.

 

Managing trusted hosts:

A list of trusted remote peers can be configured on SIP interfaces. If configured, only connections with peers in that list will be accepted. The list may contain IP-addresses or FQDNs or "remote".  If no trusted peers are configured all connections are accepted.

The SmartNode responds with a SIP " Service Unavailable” to requests received from hosts which are not trusted. 

interface sip IF_SIP
bind context sip-gateway GW_SIP
route call dest-interface IF_FXO
remote
local  
trust remote

In the example above, only requests coming from the IP address will be accepted.  Anything else will receive the " Service Unavailable" from the SmartNode.

 

In some cases, it is desirable not to send an answer back to an untrusted SIP host. For those cases, in the context sip-gateway, this behavior can be changed. By default the SmartNode responds with a “ Service Unavailable” to SIP requests received from hosts which are not trusted. With the [no] form of the command below, the SmartNode will not send any message to untrusted hosts.

[node](sip-gw)[name]# [no] answer-untrusted-hosts

context sip-gateway GW_SIP_0

interface LAN
bind interface IF_IP_LAN context router port

context sip-gateway GW_SIP_0
bind location-service LOC_SVC
no answer-untrusted-hosts
no shutdown

The SmartNode will no longer respond to untrusted hosts when the "no answer-untrusted-hosts" command is configured.

 

Frequently Asked Questions

The issue that you are experiencing might be because the DNS server does not reply at times. (this could be a firewall issue, a problem with Windows or a problem with the DNS server)

Replacing the hostname of your VoIP provider with the server IP address might help. If you do not know the IP address of the server, contact your VoIP provider, explain the issue and ask for the IP address of the server.

SIP error message might be also generated when the service you are trying to use is unavailable.

Make sure that all account details and the server hostname are entered correctly. If the issue is appearing randomly it may be temporary issue with your VoIP provider.

For additional information and assistance regarding this error message, please contact your VoIP service provider.

 

If you are using TLS as transport type, you might see this error message if there is a problem with your certificate file.

 

 

Android:

 

Android 6:

Error sometimes happens when the Android firewall suddenly decides to block access to the DNS server. 

The Zoiper log file will have a line like this:

sprers.eu

Problem:

All Endpoints from Branch locations can call HQ, but HQ is unable to call them as shown below.

Topology

Call Fails from HQ to BR1

Call Successful from BR1 to HQ

Troubleshooting:

All Clusters are configured with ILS/GPDR for Inter-site Dialing with SME as a Centralized Unit.
Step 1:
Understand/Analyze Call Flow.

Also, verify if IP Phone has visibility to Dialed or Learned Number.
NOTE: DNA doesn&#;t work in case of ILS/GDPR

Steps 2:
Make a test call to any Branch and collect the SDL logs from HQ-SUB (primary call processing node), SME, and BR1.
You can either use RTMT or CUCM CLI login.
In case, if you are using CUCM CLI use below command to capture SDL traces.
“file tail activelog cm/trace/ccm/SDL recent”

Step 3:
SIP Call Flow

This is how the Basic SIP Call Setup looks like when the calls are working properly. Now Let’s have a look at Call Flow Diagram for our scenario.

HQ-Sub

As per the logs, we are receiving SIP Response “ Service Unavailable” from SME, which is the cause of call failure.
Let’s analyze it further and have a look at SDL logs on SME.

HQ-SME

As per the logs, no SIP Request is being forwarded to BR1-Pub. It indicates that the “ Service Unavailable” was been generated by SME.

SIP Responses

Following is the snippet for SIP Response received from SME.

In our case, Subscriber is Primary Call Manager on HQ, while the IP address configured on SME to HQ SIP trunk is pointing towards HQ Publisher (Backup Call Processing node).

This is the reason Call from HQ to all Branch Clusters is failing.

Resolution:

Currently, SIP trunk on SME which is pointing towards HQ-Pub.

We need to make sure that SIP Trunk is pointing towards Primary Call Manager (HQ-Sub) and not to Secondary (HQ-Pub) as below.

Reset the SIP trunk once the destination address is modified.

Verification:

Make a test call from HQ to BR 1.

Call between HQ to all BR is successful.

Summary:

In our scenario depicted above HQ-Sub is a primary call processing node. The primary call processing node has the Cisco Call Manager Service enabled. Devices such as phones, gateways, and media resources can register and make calls only to servers with this service enabled.

Whereas IP address configured on the SME to HQ SIP Trunk is pointing towards HQ Publisher which is not a primary call processing server, whereas the Source IP address that is sending the SIP INVITE is of HQ Subscriber and it does not exist in the SIP Trunk in CUCM. This is the reason behind the Service Unavailable.

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Problem:

All Endpoints from Branch locations can call HQ, but HQ is unable to call them as shown below.

Topology

Call Fails from HQ to BR1

Call Successful from BR1 to HQ

Troubleshooting:

All Clusters are configured with ILS/GPDR for Inter-site Dialing with SME as a Centralized Unit.
Step 1:
Understand/Analyze Call Flow.

Also, verify if IP Phone has visibility to Dialed or Learned Number.
NOTE: DNA doesn&#;t work in case of ILS/GDPR

Steps 2:
Make a test call to any Branch and collect the SDL logs from HQ-SUB (primary call processing node), sip error 503, SME, and BR1.
You can either use RTMT or CUCM CLI login.
In case, sip error 503, if you are using CUCM CLI use below command to capture SDL traces.
“file tail activelog cm/trace/ccm/SDL recent”

Step 3:
SIP Call Flow

This is how the Basic SIP Call Setup looks like when the calls are working properly, sip error 503. Now Let’s have a look at Call Flow Diagram for our scenario.

HQ-Sub

As per the logs, we are receiving SIP Response “ Service Unavailable” from SME, which is the cause of call failure.
Let’s analyze it further and have a look at SDL logs on SME.

HQ-SME

As per the logs, no SIP Request is being forwarded to BR1-Pub. It indicates that the “ Service Unavailable” was been generated by SME.

SIP Responses

Following is the snippet for SIP Response received from SME.

In our case, Subscriber is Primary Call Manager on HQ, while the IP address configured on SME to HQ SIP sip error 503 is pointing towards HQ Publisher (Backup Call Processing node).

This is the reason Call from HQ to all Branch Clusters is failing.

Resolution:

Currently, SIP trunk on SME which is pointing towards HQ-Pub.

We sip error 503 to make sure that SIP Trunk is pointing towards Primary Call Manager (HQ-Sub) and not to Secondary (HQ-Pub) as below.

Reset the SIP trunk once the destination address is modified.

Verification:

Make a test call from HQ to BR 1.

Call between HQ to all BR is successful.

Summary:

In our scenario depicted above HQ-Sub is a primary call processing node. The primary call processing node has the Cisco Call Manager Service enabled. Devices such as phones, gateways, and media resources can register and make calls only to servers with this service enabled.

Whereas IP address configured on the SME to HQ SIP Trunk is pointing towards HQ Publisher which is not a primary call processing server, whereas the Source IP address that is sending the SIP INVITE is of HQ Subscriber and it does not exist in the SIP Trunk in CUCM. This is the reason behind the Service Unavailable.

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SIP Error Response = Service Unavailable

SIP Error Response = Service Unavailable

Bonker(TechnicalUser)

(OP)

Hi All,
I have a issue, I have an IP Office connected via SIP to Session Manager. When I make a call from CM to the IP Sip error 503 i get the below error Response = Service Unavailable.

The the trunks on the IP Office is configure as SMLine.


2/13/15 PMms Line = 9, Channel = 1, sip error 503, SIP Message = Response, Direction = From Switch, From = [email protected], To = @sprers.eu, Response = Service Unavailable
2/13/15 PMms Line = 9, Channel = 1, sip error 503, SIP Message = Ack, Direction = To Switch, From = [email protected], To = @sprers.eu

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"SIP/ Service Unavailable" error message when you make a PSTN call in Lync Server

Symptoms

Consider the following scenario:

  • A user makes a Public Switched Telephone Network (PSTN) call In a Microsoft Sip error 503 Server environment.

  • The PSTN call is routed from a Mediation server to a gateway that is disconnected from the Mediation server.

In this scenario, the Mediation server rejects the call, sip error 503. Additionally, the Mediation server returns the following error message:

SIP/ Service Unavailable


Note This error message only contains an "ms-trunking-peer-state" header. However, it should also contain an "ms-trunking-peer" header. 

Resolution

To resolve this issue, install the following cumulative update:

Description of the cumulative update for Lync ServerMediation server: April

Status

Microsoft has confirmed that this is a problem in the Microsoft products that are listed in the "Applies to" section.

Some other error (1): Operation not permitted

 

As a workaround, try the Settings screen and select Data usage. You can tap a specific application and check the "Restrict mobile data" check box to prevent the application from using mobile data in the background.

 

 

 

No route to destination

This error usually means the network is not active yet. Try if you can use a browser to search for something on Google. Most likely that will not work either. 

Try to fix the network first, then try to re-register in Zoiper.

 

No transports left to try

Android:

In rare cases this might occur after a network change, sip error 503. The issue seems that Android signals a network change before the network is completely up and as a result the DNS lookups fail.  We are trying to work around this android specific race condition.

When you see this error on Android, please contact us on [email protected] so that we can help to troubleshoot the issue.

 

 

Transport failure

When you see this error on Android, please contact us on s[email protected] so sip error 503 we can help to troubleshoot the issue

 

DNS Timeout

 When you see this sip error 503 on Android, please contact us on [email protected] so that we can help to troubleshoot the issue.

 

Outbound calls error with "all circuits busy" or "congestion":

This is the default configuration of Asterisk regardless of the actual error generated (which is infuriating when you are trying to diagnose the real problem) unless PBX is updated to send back the real error rather than the changed error. This error most commonly occurs when the call is not authenticating properly, at which point check the above in the SIP trunk configuration sip error 503 Asterisk, sip error 503, swap username= for defaultuser= to see if this solves the issue. Just because a trunk is showing as registered does not mean it will authenticate correctly.

Outbound calls fail with SIP error (Not Accepted Here) or I-SUP errors 58 (bearer capability not available) or 88 (incompatible destination):

Check the codecs allowed in the SIP trunk configuration above, VoiceHost only supports: alaw, ulaw, gsm
If a codec is defined in Asterisk that is not one of the above, or is offering a differing sample rate or interval rate (e.g. /20i - Hz at 20ms) cannot interwork with /30i - Hz at 30ms) the call will fail and the codecs in the SIP trunk configuration need to be aligned to use one of the above codecs.

Inbound calls fail with SIP error (Request Timeout):

Check the inbound number is mapped in the system correctly, if necessary the SIP trunk on ora-00600 internal error code, arguments [17126] portal can be configured to strip the plus, e.g. if Asterisk is configured to use plus somewhere else. Check the trunk is registered. Ascertain how long the error took to come back if it was immediate the trunk is usually unregistered if it took a few seconds the number is usually not mapped correctly.

Calls fail with SIP errorI-SUP errors 34 or

If our platform replies back with it usually means the gateway trying to process the call can't due to "issues", or the customer has hit their Calls-Per-Second (CPS) limit and is sending too many calls at once. Sometimes the error is passed back from IP Exchange through VoiceHost to the customer's system, at which point the call will usually hunt to another route to try and place the call.

Cause code (ISUP)SIP EquivalentDefinition
1 Not FoundUnallocated (unassigned) number
2 Not foundno route to network
3 Not foundno route to destination
16BYE or CANCEL (*)normal call clearing
17 Busy hereuser busy
18 Request Timeoutno user responding
19 Temporarily unavailableno answer from the user
20 Temporarily unavailablesubscriber absent
21 Forbidden (+)call rejected
22 Gonenumber changed (w/o diagnostic)
22 Moved Permanentlynumber changed (w/ diagnostic)
23 Goneredirection to new destination
26 Not Found (=)non-selected user clearing
27 Bad Gatewaydestination out of order
28 Address incompleteaddress incomplete
29 Not implementedfacility rejected
31 Temporarily unavailablenormal unspecified
34 Service unavailableno circuit available
38 Service unavailablenetwork out of order
41 Service unavailabletemporary failure
42 Service unavailableswitching equipment congestion
47 Service unavailableresource unavailable
55 Forbiddenincoming calls barred within CUG
57 Forbiddenbearer capability not authorized
58 Service unavailablebearer capability not presently
65 Not Acceptable Herebearer capability not implemented
70 Not Acceptable Hereonly restricted digital avail
79 Not implementedservice or option not implemented
87 Forbiddenuser not member of CUG
88 Service unavailableincompatible destination
Gateway timeoutrecovery of timer expiry
Server internal errorprotocol error
Server internal errorinterworking unspecified

Bria Account Error: Sip error 503 to Enable SIP Error

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Symptom

In Bria, the following error appears: Account Failed To Enable SIP Error

Bria5-siperrorPNG

Applies To

  • 8x8 Contact Center
  • Agents
  • Bria softphone

Resolution

  1. Launch Bria.
  2. Select Softphone and select Account Settings.
  3. Click Edit Account.
  4. Select the Transport tab.
  5. Deselect IPv6 checkbox.
    Bria5_sprers.eu
  6. Click OK.

Cause

The agent's OS or network is not configured for IPv6.

Additional Information

Please go here for more information on configuring Bria for 8x8 Contact Center.

1 Comments

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