You’d need to actually identify the underlying problem. Have you checked the SIP/SDP (sip set debug on) to see where it says to send media? Have you checked “rtp set debug on” to see if the media is being sent? Have you done a packet capture to examine traffic?
I’d also note that you have NAT enabled for that peer, meaning that Asterisk will send media to where media was received from - which may differ from what was in the SDP and cause a problem.
v=0
o=- 1074043495 1074043497 IN IP4 172.29.108.4
s=SBC call
c=IN IP4 172.29.108.4
t=0 0
m=audio 41320 RTP/AVP 0 116
a=rtpmap:0 PCMU/8000
a=rtpmap:116 telephone-event/8000
a=ptime:20
<------------->
— (12 headers 9 lines) —
Sending to 192.168.224.195:9133 (NAT)
Found RTP audio format 0
Found RTP audio format 116
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 116
Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.29.108.4:41320
In the SDP you’ve provided it states that Asterisk should send audio to 172.29.108.4. There is no configuration to override this, except for the nat option which will cause Asterisk to send to the source address instead.
It’s also giving a 172 address as the contact address, so you will need routing for that RFC 1918 network as well as the 192 one, or you will not be able to hang up the call from your end.
Missed that. I do not understand what purpose the 192 network serves, or what “Address exposed to Tech Services” means.
The ITSP has sent c=IN IP4 172.29.108.4 which means that Asterisk will use that as the media address without your having to do anything.
It has also sent Contact: sip:172.29.108.3:9133, which will be used as the signalling address. However it has sent the request from 192.168.224.195:9133 which is the “named [I]P address”, whatever that means.
Do you have a route t the 172 network?
As has already been said, you need to do nothing to have the correct remote addresses used when you are the UAS, as is the case here. All the addresses are communicated in the SIP request. Even when acting as UAC, you don’t need to do anything about the remote media address, as that is provided in the singalling.