(lists) Denis BUCHER
2015-Nov-12 15:22 UTC
[asterisk-users] No sound with internal calls depending on which phones
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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Mitul Limbani
2015-Nov-12 15:25 UTC
[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 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) > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151112/e2ddfc25/attachment.html>
jg
2015-Nov-12 15:57 UTC
[asterisk-users] No sound with internal calls depending on which phones
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151112/98f023e5/attachment.html>
Sam Basan
2015-Nov-12 16:05 UTC
[asterisk-users] No sound with internal calls depending on which phones
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 at enterux.in]
Sent: Thursday, November 12, 2015 5:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users
at lists.digium.com>
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
On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" <dbucherml at
hsolutions.ch <mailto: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 <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 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
<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)
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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