Hi folks, I'm looking for some basic troubleshooting help. I have a fairly basic home setup running astlinux-1.4.3 x86_64 - Asterisk 13.38.2. I have a couple of Cisco CP-7811 phones and a couple Avaya J129s. I use Voip.ms as my provider.
At some point recently, I noticed that when making an outgoing call, my Cisco phones were dropping their outgoing audio within a half-second of making a connection. My incoming audio is fine. Everything is fine with incoming calls. I then tried out my Avaya phones and they have no issues under any circumstances. So it seems like I need to make a change to my Cisco phones, but I just have no idea what. These logs probably aren't detailed enough, but I'll start with them for now. There are only 2 differences which I've highlighted, otherwise the logs are the same: More info to help reading below: My 'home' number aka Asterisk: 6137778888 Internal extensions: 200 Cisco - 192.168.2.147 Internal extensions: 400 Avaya - 192.168.2.157 My external cell number for testing: 3439998888 Voip.ms server: 208.100.60.50 Here is the Cisco phone (ext 200) calling my cell phone == Using SIP RTP CoS mark 5 > 0x152658048950 -- Strict RTP learning after remote address set to: 192.168.2.147:16412 -- Executing [3439998888@default:1] Set("SIP/200-000000d9", "CALLERID(all)=LAW <6137778888>") in new stack -- Executing [3439998888@default:2] Dial("SIP/200-000000d9", "SIP/3439998888@voipms") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/3439998888@voipms > 0x152664007350 -- Strict RTP learning after remote address set to: 208.100.60.50:17166 -- SIP/voipms-000000da is making progress passing it to SIP/200-000000d9 > 0x152658048950 -- Strict RTP switching to RTP target address 192.168.2.147:16412 as source > 0x152664007350 -- Strict RTP switching to RTP target address 208.100.60.50:17166 as source ****** This line not in the other log ******* > 0x152658048950 -- Strict RTP learning complete - Locking on source address 192.168.2.147:16412 -- SIP/voipms-000000da answered SIP/200-000000d9 -- Channel SIP/voipms-000000da joined 'simple_bridge' basic-bridge <7dfd9292-27b5-4c07-92a8-33d435191096> -- Channel SIP/200-000000d9 joined 'simple_bridge' basic-bridge <7dfd9292-27b5-4c07-92a8-33d435191096> > Bridge 7dfd9292-27b5-4c07-92a8-33d435191096: switching from simple_bridge technology to native_rtp > Remotely bridged 'SIP/200-000000d9' and 'SIP/voipms-000000da' - media will flow directly between them > 0x152664007350 -- Strict RTP learning complete - Locking on source address 208.100.60.50:17166 -- Channel SIP/200-000000d9 left 'native_rtp' basic-bridge <7dfd9292-27b5-4c07-92a8-33d435191096> -- Channel SIP/voipms-000000da left 'native_rtp' basic-bridge <7dfd9292-27b5-4c07-92a8-33d435191096> == Spawn extension (default, 3439998888, 2) exited non-zero on 'SIP/200-000000d9' Here's the Avaya (400) doing the same call: == Using SIP RTP CoS mark 5 > 0x1526800401c0 -- Strict RTP learning after remote address set to: 192.168.2.157:5004 -- Executing [3439998888@default:1] Set("SIP/400-000000db", "CALLERID(all)=LAW <6137778888>") in new stack -- Executing [3439998888@default:2] Dial("SIP/400-000000db", "SIP/3439998888@voipms") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/3439998888@voipms > 0x152674006650 -- Strict RTP learning after remote address set to: 208.100.60.50:15962 -- SIP/voipms-000000dc is making progress passing it to SIP/400-000000db > 0x1526800401c0 -- Strict RTP switching to RTP target address 192.168.2.157:5004 as source > 0x152674006650 -- Strict RTP switching to RTP target address 208.100.60.50:15962 as source > 0x1526800401c0 -- Strict RTP learning complete - Locking on source address 192.168.2.157:5004 ****** This line not in the other log ******* -- SIP/voipms-000000dc requested media update control 26, passing it to SIP/400-000000db -- SIP/voipms-000000dc answered SIP/400-000000db -- Channel SIP/voipms-000000dc joined 'simple_bridge' basic-bridge <4516067d-beca-43e1-b92f-78def4c48c4e> -- Channel SIP/400-000000db joined 'simple_bridge' basic-bridge <4516067d-beca-43e1-b92f-78def4c48c4e> > Bridge 4516067d-beca-43e1-b92f-78def4c48c4e: switching from simple_bridge technology to native_rtp > Remotely bridged 'SIP/400-000000db' and 'SIP/voipms-000000dc' - media will flow directly between them -- Channel SIP/voipms-000000dc left 'native_rtp' basic-bridge <4516067d-beca-43e1-b92f-78def4c48c4e> -- Channel SIP/400-000000db left 'native_rtp' basic-bridge <4516067d-beca-43e1-b92f-78def4c48c4e> == Spawn extension (default, 3439998888, 2) exited non-zero on 'SIP/400-000000db' I appreciate any and all help! Craig
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