[asterisk-users] No sound with internal calls depending on which phones
Ishfaq Malik
ish at pack-net.co.uk
Thu Nov 12 10:50:54 CST 2015
That is correct for turning SRTP off on a Snom phone.
On 12 November 2015 at 16:46, (lists) Denis BUCHER <dbucherml at 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 ?
>
> 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,*
>
> [image: cid:image001.jpg at 01D0D5C4.27A0CBA0]
>
> *Sam Basan*
>
> [image: cid:image003.png at 01C918DA.6B3E4530]
>
>
>
> *From:* Mitul Limbani [mailto:mitul at enterux.in <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> <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>
> 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>
> 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
>
>
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151112/4e6f0185/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 2008 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151112/4e6f0185/attachment.jpe>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/png
Size: 2038 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151112/4e6f0185/attachment.png>
More information about the asterisk-users
mailing list