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

[Lonnie A. <li...@lo...>]

Hi Craig,

To test, in /etc/asterisk/rtp.conf try setting "strictrtp=no" instead of the default "yes".  Read the comments why this is enabled by default.  Restart Asterisk and see if it helps.  If not return it back.

I have seen occasions where "strictrtp=yes" has caused issues.

Lonnie


> On Oct 16, 2021, at 2:13 PM, Craig Law <cra...@gm...> wrote:
> 
> 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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