No audio on outgoing calls

Asterisk Asterisk Support
1

I recently moved my customer to a new location and because the could not get a static IP right away we moved their existing SIP trunks to my PBX and routed all call in and out of my system. When they received their static IP we the trunks to their system.

They can receive incoming calls with no problems but when they try to make outgoing calls there is no audio. What we see when we run a trace we see their system sending multiple invites. The 183 Session progress message comes in and then the audio but it never makes it to the phone.

It appears that the device is not registering with the proxy. This system was working fine in the previous location.

Trace output
12 4.273427 192.168.10.252 192.168.1.60 SIP/SDP Request: INVITE sip:xxxxxxxxxxx@192.168.1.60:5060, with session description
13 4.274213 192.168.10.1 192.168.10.252 SIP Status: 407 Proxy Authentication Required
14 4.340840 192.168.10.252 192.168.1.60 SIP Request: ACK sip:xxxxxxxxxxx@192.168.1.60:5060
15 4.356206 192.168.10.252 192.168.1.60 SIP/SDP Request: INVITE sip:xxxxxxxxxxx@192.168.1.60:5060, with session description
16 4.358050 192.168.10.1 192.168.10.252 SIP Status: 100 Trying
17 4.599880 192.168.1.60 216.82.224.202 SIP/SDP Request: INVITE sip:+1xxxxxxxxxx@216.82.224.202, with session description
18 4.677032 216.82.224.202 192.168.1.60 SIP Status: 100 Giving a try
19 5.599344 192.168.1.60 216.82.224.202 SIP/SDP Request: INVITE sip:+1xxxxxxxxxx@216.82.224.202, with session description
20 5.674068 216.82.224.202 192.168.1.60 SIP Status: 100 Giving a try
21 6.397207 216.82.224.202 192.168.1.60 SIP/SDP Status: 183 Session Progress, with session description
22 6.426208 63.215.29.79 192.168.1.60 RTP PT=ITU-T G.711 PCMU, SSRC=0x6BEC7CE2, Seq=64350, Time=1042974671, Mark
23 6.444371 63.215.29.79 192.168.1.60 RTP PT=ITU-T G.711 PCMU, SSRC=0x6BEC7CE2, Seq=64351, Time=1042974831

…

Any ideas?

2

Ok, here is the resolution to this issue. I would have not believed that it fixed the issue unless I had not made the change myself. I even paid for support from xxxxxxxxx and did not get the resolution from them.

After 1 week of going back and forth with support and trying this and that I decided I would add the following line to sip_nat.conf

fromdomain=xxxxxxxx.com

Since my customer doe not have a domain name I created one on dyndns.com and entered that in the fromdomain line. I restarted Asterisk and everything worked correctly. This system had been up and running since October of 2008 and I had never had a domain name in the file. Since the customer had to change ISPs that is where the issue came up. The ISP for this customer is Windstream. I have not been able to get any info from Windstream as to why this is needed from their perspective.

I hope this helps other that are experiencing this issue.