[Astlinux-users] One-way audio problem outgoing on Cisco phone but not Avaya

[Craig L. <cra...@gm...>]

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