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 <craigsteven...@gmail.com> 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
>
> _______________________________________________
> Astlinux-users mailing list
> Astlinux-users@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> pay...@krisk.org.
_______________________________________________
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
pay...@krisk.org.
