Hello,
Here you are a description of the environment:
SIP Trunk Early Offer (without MTP checked) between Cisco CallManager v9.1 and an ip PBX Asterisk 1.6 based.
We have resume issue. If a cisco phone place on hold an ip phone (Yealink) registered with Asterisk it works fine.
If i resume from hold i have one-way audio. We see SIP traces from RTMT and notice that Cisco CallManager advertises Asterisk with new SIP SDP port,
but ip phone registered with asterisk still sends RTP packets to the old port previously negotiated.
We also saw the hold\resume SIP handshake between Yealink ip phones (all phones registered with PBX Asterisk) and observed new negotiated SDP port
is the same as the previous one.
Is there is a way for CISCO CallManager to nogotiate the same RTP port when resume a call from hold?
nextis the SIP trace:
172.23.112.10 = PBX Asterisk
172.23.112.20 = Cisco CM
172.23.112.194 = Yealink ip phone (not displayed)
172.23.112.23 = cisco ip phone (not displayed)
messages from [1]to[9] are related to the first handshake - when cisco ip phone invites Yealink. RTP is flowing end-to-end between end devices.
messages from [10]to[13] are related to the place on hold - cisco places Yealink on hold.
messages from [14]to[21] are related to resume - cisco CM include a new RTP port in SDP ACK[17], but this new characteristic of the media stream
seems to be ignored by Asterisk.
Thanks
regards
