Did you get this SIP error on your Cisco call manager?
You already change all the codec, configured transcoder which is register with call manager and everything seems fine. But outgoing calls still give this error!!!
Via: SIP/2.0/UDP 192.168.1.1:5060;received=192.168.1.1;branch=fg5df4bK4F1C77
CSeq: 101 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Based on the Wikipedia article List of SIP response codes
488 Not Acceptable Here
Some aspect of the session description or the Request-URI is not acceptable, or Codec issue.
If we want to go to more deep, the IETF rfc3261 explain about that message:
488 Not Acceptable Here
The response has the same meaning as 606 (Not Acceptable), but only
applies to the specific resource addressed by the Request-URI and the
request may succeed elsewhere.
A message body containing a description of media capabilities MAY be
present in the response, which is formatted according to the Accept
header field in the INVITE (or application/sdp if not present), the
same as a message body in a 200 (OK) response to an OPTIONS request.
Often this is related to codec incompatibilities. For anyone encountering this issue, they should check whether both sides (server and client) have at least one codes they can negotiate.
For fix that you should check both side configuration, I mean if you got an error for a SIP trunk you should check the SIP trunk provider codec and config if it’s interconnection in your company you should check the second server configuration.
For finding and debugging the codec for both side, you can capture SIP session by tools like Wireshark; you can check the codec for each call.
To fix it you should go to SIP trunk and look for “MTP Preferred Originating Codec” and change it. Make sure both sides use the same codec.
Login into your Cisco Unified CM Administration and click on Device menu.
Click on Trunk
Select your SIP trunk and click on to change the configuration.
In SIP trunk configuration goto “SIP Information” section and check the value of “MTP Preferred Originating Codec.”
If the problem is still unresolved, there is one more step.
Most of SIP provider want Early Offer INVITEs. They use this always to decide on which codec to offer for the calls.
To configure Delayed-Offer to Early-Offer for SIP Audio Calls at the global level, perform the steps in this section.
Open a terminal and connect to your CUCM console.
And enter the following commands:
2. configure terminal
3. voice service voip
4. allow-connections sip
5. early-offer forced
Latest posts by Reza Mousavi (see all)
- How To Create Yeastar S-Series Dashboard In 10 Minutes - 2019-03-21
- How RTP (Real-time Transport Protocol ) Works in VOIP? - 2019-03-18
- Sample Report – Top 25 Trunks/COs – Incoming - 2019-03-15