[asterisk-users] No sound with internal calls depending on which phones

[Mitul Limbani]

You might have to disable srtp negotiations inside the phone web ui
options.

Mitul
On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" <dbucherml at hsolutions.ch>
wrote:

> Dear all,
>
> I have a very strange problem :
>
>    - external calls work perfectly,
>    - internal calls between some phones too,
>    - but internal call between two similar phones don't work !!! (Snom
>    710)
>
> When we have sound, there are no errors in asterisk. When we do not have
> sound, there is the following error :
>
>    - [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
>    module loaded, can't setup SRTP session.
>
> This is a working internal call :
>
>   == Using SIP RTP CoS mark 5
>     -- Executing [301 at local:1] Dial("SIP/dbucher-00000000", "SIP/phone1")
> in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called phone1
>     -- SIP/phone1-00000001 is ringing
>     -- SIP/phone1-00000001 is ringing
>     -- SIP/phone1-00000001 is ringing
>     -- SIP/phone1-00000001 is ringing
>     -- SIP/phone1-00000001 is ringing
>     -- SIP/phone1-00000001 answered SIP/dbucher-00000000
>     -- Remotely bridging SIP/dbucher-00000000 and SIP/phone1-00000001
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Got  RTP packet from    192.168.128.99:49646 (type 126, seq 031575, ts
> 000001, len 000000)
> [Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read:
> Unknown RTP codec 126 received from '192.168.128.99:49646'
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
>   == Spawn extension (local, 301, 1) exited non-zero on
> 'SIP/dbucher-00000000'
>
> This is a non-working call :
>
>   == Using SIP RTP CoS mark 5
> [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
> module loaded, can't setup SRTP session.
>     -- Executing [301 at local:1] Dial("SIP/hsolutionspf5-00000002",
> "SIP/phone1") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called phone1
>     -- SIP/phone1-00000003 is ringing
>     -- SIP/phone1-00000003 is ringing
>     -- SIP/phone1-00000003 is ringing
>     -- SIP/phone1-00000003 is ringing
>     -- SIP/phone1-00000003 is ringing
>     -- SIP/phone1-00000003 answered SIP/hsolutionspf5-00000002
>     -- Remotely bridging SIP/hsolutionspf5-00000002 and SIP/phone1-00000003
> Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)
>   == Spawn extension (local, 301, 1) exited non-zero on
> 'SIP/hsolutionspf5-00000002'
>
> I tried many options to disable SRTP but without success :
>
>    - canreinvite = no
>    - canreinvite = nonat
>    - srtpcapable=no
>    - encryption=no
>    - directmedia=nonat
>    - ...or noload => res_srtp.so in modules.conf
>
>
> Any help would be GREATLY appreciated !
>
> Denis
>
> P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final)
>
>
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