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

[jg]

Am 12.11.2015 um 16:22 schrieb (lists) Denis BUCHER:
> 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)
>
>
>
Please check
http://wiki.snom.com/wiki/index.php/Settings/user_srtp
and make sure the flag is off.

If you install Asterisk with the srtp module, then you need to set the auth-tag to AES-80, but I 
haven't played with this option for quite some time.

jg
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