Did you get this SIP error on your Cisco call manager?
You already change all the codec, configured the transcoder, register with the call manager, and everything seems fine. But outgoing calls still give this error!!!
SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.1.1:5060;received=192.168.1.1;branch=fg5df4bK4F1C77 To: <sip:firstname.lastname@example.org>;tag=5435345-43543 From: <sip:email@example.com>;tag=E435A43-A43 Call-ID: FDF74C45-3C7351E2-64E3FEC8-F5473FD6@192.168.1.1 CSeq: 101 INVITE Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Contact: <sip:firstname.lastname@example.org:5060> Reason: Q.850;cause=65 Content-Length: 0</sip:email@example.com:5060></sip:firstname.lastname@example.org></sip:email@example.com>
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 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.
To fix that, you should check both side configurations; I mean, if you got an error for a SIP trunk, you should check the SIP trunk provider codec and config; if it’s an interconnection in your company, you should check the second server configuration.
For finding and debugging the codec for both sides, you can capture SIP sessions by tools like Wireshark; you can check the codec for each call.
To fix it, you should go to the SIP trunk and look for “MTP Preferred Originating Codec,” and change it. Make sure both sides use the same codec.
Login to your Cisco Unified CM Administration and click on the Device menu.
Click on Trunk
Select your SIP trunk and click on it to change the configuration.
In the SIP trunk configuration, go to the “SIP Information” section and check the value of “MTP Preferred Originating Codec.”
If the problem is still unresolved, there is one more step.
Most SIP providers 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:
1. enable 2. configure terminal 3. voice service voip 4. allow-connections sip 5. early-offer forced 6. exit
Do you want to know how your phone system is performing?
If you are looking for a way to monitor the performance of your Cisco Call Manager, then look no further! We provide everything you need in one simple package that will be easy to install and use. There’s no other service like it out there! It’s not just an amazing product but also an incredible experience you can have every day of your life.