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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151112/bdf03ccf/attachment.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2008 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151112/bdf03ccf/attachment.jpg> -------------- next part -------------- A non-text attachment was scrubbed... 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(lists) Denis BUCHER
2015-Nov-12 16:46 UTC
[asterisk-users] No sound with internal calls depending on which phones
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 ?crit :> > 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) > > > -- > _____________________________________________________________________ > -- 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/20151112/8ba9063c/attachment-0001.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/20151112/8ba9063c/attachment-0001.jpe> -------------- next part -------------- A non-text attachment was scrubbed... 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Ishfaq Malik
2015-Nov-12 16:50 UTC
[asterisk-users] No sound with internal calls depending on which phones
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 ?crit : > > 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... 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Pete Mundy
2015-Nov-12 19:36 UTC
[asterisk-users] No sound with internal calls depending on which phones
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 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 ?-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151113/c53682b5/attachment.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4118 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151113/c53682b5/attachment.bin>