[asterisk-users] No sound with internal calls depending on which phones
(lists) Denis BUCHER
dbucherml at hsolutions.ch
Thu Nov 12 10:46:35 CST 2015
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 :
>
> Snom default configuration is SRTP enabled.
>
> You should disable the SRTP from the phone web GUI configuration
>
> **
>
> **
>
> *Sincerely,*
>
> cid:image001.jpg at 01D0D5C4.27A0CBA0
>
> *Sam Basan*
>
> cid:image003.png at 01C918DA.6B3E4530
>
> *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)
>
>
> --
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