SIP Call Flow - Invite from ISP after RTP established
I have the following issue: when I try to dial a PSTN number, from the customer, using the G729 the call is not established and I'm getting a fast busy. The ISP is telling us that we should be sending a 488 after their Invite w/SDP.
What I find strange is that after the RTP seems to be established there is a new INVITE SDP from the ISP requesting the call to be done in g729 annexb.
Here's the call flow (220.127.116.11 is the Customer and 18.104.22.168 the IMS Provider):
You need to go back to them. "488 means media unacceptable", I am not sure they expect you to send that after a RE-INVITE. They shouldn't send a 200 OK to your INVITE if they cant do g729. You may also want to offer them G729 annexb if thats the only flavor of g729 they support
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
SIP traces provide key information in troubleshooting SIP Trunks, SIP
endpoints and other SIP related issues. Even though these traces are in
clear text, these texts can be gibberish unless you understand fully
what they mean. This document attempts to br...
Please find the attached HTML document, download and open it on your PC.
This provides an easy to use form where you simply answer a few
questions and it will render the proper jabber-config.xml file for you
to copy/paste. There is built in logic to verif...
[toc:faq]CUCM Database Replication is an area in which Cisco customers
and partners have asked for more in-depth training in being able to
properly assess a replication problem and potentially resolve an issue
without involving TAC. This document discusse...