No Sound With Internal Calls Depending On Which Phones

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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 :
This is a non-working call :
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)

6 thoughts on - No Sound With Internal Calls Depending On Which Phones

  • ——=_NextPart_001_03BE_01D11D74.BC99E870
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    Snom default configuration is SRTP enabled.

    You should disable the SRTP from the phone web GUI configuration

    Sincerely,

    Sam Basan

    From: Mitul Limbani [mailto:mitul@enterux.in]
    Sent: Thursday, November 12, 2015 5:25 PM
    To: Asterisk Users Mailing List – Non-Commercial Discussion
    Subject: Re: [asterisk-users] No sound with internal calls depending on which phones

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

    Mitul

    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@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 <http://192.168.128.99:49646> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646> (type 00, len 000160)
    Got RTP packet from 192.168.128.99:49646 <http://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 <http://192.168.128.99:49646> ‘
    Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.231:57818 <http://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@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 <http://192.168.128.228:65494> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.228:65494 <http://192.168.128.228:65494> (type 00, len 000160)
    Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350> (type 03, len 000033)
    Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350> (type 03, len 000033)
    Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350> (type 03, len 000033)
    Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350> (type 03, len 000033)
    Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350> (type 03, len 000033)
    Sent RTP P2P packet to 192.168.128.231:51350 <http://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)

  • Dear Sam, dear jg, dear Mitul, dear all,

    Thanks a lot for your advices! I had the same idea, but it still doesn’t work!

    Maybe I changed the wrong option on the GUI configuration ?
    I went to menu “Setup” > “Identity 1” > “RTP” > “RTP Encryption:” >
    “off” on both phones. And in the configuration I see “user_srtp1!: off”

    Is this right ?

    Denis

    Le 12.11.2015 17:05, Sam Basan a écrit :

  • –001a114b0c4e10509205245abf67
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    That is correct for turning SRTP off on a Snom phone.

  • –Apple-Mail=_D308EE46-346A-4B70-84BF-7292D9A07CAC
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    charset=us-ascii

    Hi Denis

    That advice is correct for disabling RTP support in the phone and if you have achieved this then your quoted error about SRTP in the Asterisk console (when the call is failing) should no longer be appearing.

    This will help show that it was a red herring all along.

    The next step (IMO) is to use the Snom’s built-in packet capture capabilities to grab a packet capture of a failed conversation from each phone then post it somewhere with a link to the list so that others can inspect the SIP signalling to discover where the issue lies.

    You may also need to provide some information about your network configuration, IP ranges, firewall etc (a little diagram goes a long way).

    For information on how to use the packet capture capabilities on the phone refer the Snom user’s guide. I’m pretty sure it’s well documented.

    Hope this helps and look forward to investigating the packet captures for you 🙂

    Pete

    –Apple-Mail=_D308EE46-346A-4B70-84BF-7292D9A07CAC
    Content-Transfer-Encoding: quoted-printable Content-Type: text/html;
    charset=utf-8

    Hi Denis

    That advice is correct for disabling RTP support in the phone and if you have achieved this then your quoted error about SRTP in the Asterisk console (when the call is failing) should no longer be appearing.
    This will help show that it was a red herring all along.
    The next step (IMO) is to use the Snom’s built-in packet capture capabilities to grab a packet capture of a failed conversation from each phone then post it somewhere with a link to the list so that others can inspect the SIP signalling to discover where the issue lies.
    You may also need to provide some information about your network configuration, IP ranges, firewall etc (a little diagram goes a long way).
    For information on how to use the packet capture capabilities on the phone refer the Snom user’s guide. I’m pretty sure it’s well documented.
    Hope this helps and look forward to investigating the packet captures for you 🙂
    Pete

    On 13/11/2015, at 5:46 AM, (lists) Denis BUCHER <dbucherml@hsolutions.ch> wrote:


    Dear Sam, dear jg, dear Mitul, dear
    all,

    Thanks a lot for your advices! I had the same idea, but it still
    doesn’t work!

    Maybe I changed the wrong option on the GUI configuration ?
    I went to menu “Setup” > “Identity 1” > “RTP” > “RTP
    Encryption:” > “off” on both phones.
    And in the configuration I see “user_srtp1!: off”

    Is this right ?