Re: [asterisk-users] SIP trunks going to the wrong context
On Thursday, December 14, 2017, 10:05:23 PM, Tony Mountifield (t...@softins.co.uk) wrote: > So I think you really do need to have a single peer section for all sipgate > calls, pointing to one sipgate context in your dialplan that contains all > your various extensions like se2489, sj0151, etc. That is what I do - all my incoming calls (which originate from the same IP and Port) go into a single context in extensions.conf, from where they are directed into individual call handlers depending on the DCID [incoming calls] exten => 4420,n,Goto(handler-a,s,1) exten => 4400,n,Goto(handler-b,s,1) exten => 4421,n,Goto(handlet-c,s,1) exten => 4422,n,Goto(handler d,s,1) -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunks going to the wrong context
In article <1513290358.2926.4.ca...@linuxista.com>, Frank Vanoni wrote: > I don't know if it applies to your problem, but I also had some > troubles with multiple account on same SIP provider. > Here what works for me: > > > In sip.conf: > > > register => 11:qwe...@sip.provider.zz/11 ; Trunk1 > register => 22:asd...@sip.provider.zz/22 ; Trunk2 > register => 22:yxc...@sip.provider.zz/22 ; Trunk3 > > > [trunk1] > type=friend > host=sip.provider.zz > defaultuser=11 > secret=qwertz > canreinvite=no > insecure=invite > nat=force_rport,comedia > qualify=yes > context=trunkincoming > description=Trunk 1 > > [trunk2] > type=friend > host=sip.provider.zz > defaultuser=22 > secret=asdfgh > canreinvite=no > insecure=invite > nat=force_rport,comedia > qualify=yes > context=trunkincoming > description=Trunk 2 > > [trunk3] > type=friend > host=sip.provider.zz > defaultuser=33 > secret=yxcvbn > canreinvite=no > insecure=invite > nat=force_rport,comedia > qualify=yes > context=trunkincoming > description=Trunk 3 > > > > In extensions.conf: > > [trunkincoming] > exten => 11,1,GoTo(firstline,11,1) > exten => 22,1,GoTo(secondline,22,1) > exten => 33,1,GoTo(thirdline,33,1) > > [firstline] > exten => 11,1,Dial(SIP/officephone,120,m) > > [secondline] > exten => 22,1,Dial(SIP/livingroomphone,120,m) > > [thirdline] > exten => 33,1,Dial(SIP/bedroomphone,120,m) But because you have all three of your trunk peers pointing to the same context, you don't necessarily know whether the inbound calls are matching different peers or all the same one. If you had each trunk pointing at a different context, you would probably run into the same problem as the OP. Cheers, Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunks going to the wrong context
I don't know if it applies to your problem, but I also had some troubles with multiple account on same SIP provider. Here what works for me: In sip.conf: register => 11:qwe...@sip.provider.zz/11 ; Trunk1 register => 22:asd...@sip.provider.zz/22 ; Trunk2 register => 22:yxc...@sip.provider.zz/22 ; Trunk3 [trunk1] type=friend host=sip.provider.zz defaultuser=11 secret=qwertz canreinvite=no insecure=invite nat=force_rport,comedia qualify=yes context=trunkincoming description=Trunk 1 [trunk2] type=friend host=sip.provider.zz defaultuser=22 secret=asdfgh canreinvite=no insecure=invite nat=force_rport,comedia qualify=yes context=trunkincoming description=Trunk 2 [trunk3] type=friend host=sip.provider.zz defaultuser=33 secret=yxcvbn canreinvite=no insecure=invite nat=force_rport,comedia qualify=yes context=trunkincoming description=Trunk 3 In extensions.conf: [trunkincoming] exten => 11,1,GoTo(firstline,11,1) exten => 22,1,GoTo(secondline,22,1) exten => 33,1,GoTo(thirdline,33,1) [firstline] exten => 11,1,Dial(SIP/officephone,120,m) [secondline] exten => 22,1,Dial(SIP/livingroomphone,120,m) [thirdline] exten => 33,1,Dial(SIP/bedroomphone,120,m) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunks going to the wrong context
In article , Ade Vickers wrote: > Hi all, > > I'm trying to resolve a weird issue with SIP routing. > > I have a number of SIP trunks, from a selection of providers, all of > which are registered in sip.conf: > > [general] > context=default > allowguest=no > allowoverlap=no > udpbindaddr=0.0.0.0 > tcpenable=yes > tcpbindaddr=0.0.0.0 > transport=udp > bindport=15060 > srvlookup=yes > allowsubscribe=yes > limitonpeers=yes > callcounter=yes > vmexten=5199 > nat=no > > ; SE registrations > register => user1:passwo...@sipgate.co.uk:5060/se2489 > register => user2:passwo...@sipgate.co.uk:5060/se1268 > register => user3:passwo...@sipgate.co.uk:5060/se0845 > register => user4:passwo...@callcentric.com:5060/se1777 > register => user5:passwo...@sipgate.co.uk:5060/se4130 > register => user9:passwo...@sip.vohippo.com:5060/se1413 > > ; SJ registrations > register => user6:passwo...@sipgate.co.uk:5060/sj0151 > register => user7:passwo...@callcentric.com:5060/sj1777 > register => user8:passwo...@sipgate.co.uk:5060/sj0203 > > I then have a selection of #included files. The first defines se2489: > > [se2489] > type=friend > insecure=port,invite > secret=password1 > defaultuser=user1 > fromuser=user1 > fromdomain=sipgate.co.uk > host=sipgate.co.uk > port=5060 > outboundproxy=proxy.live.sipgate.co.uk > disallow=all > allow=ulaw > context=external-se > qualify=yes > canreinvite=no > dtmfmode=rfc2833 > > The second defines sj0151: > > [sj0151] > type=friend > insecure=port,invite > secret=password6 > defaultuser=user6 > fromuser=user6 > fromdomain=sipgate.co.uk > host=sipgate.co.uk > outboundproxy=proxy.live.sipgate.co.uk > disallow=all > allow=ulaw > context=sj-main > regcontext=sj-main     ; Added to try to fix wrong context on IB > calls > subscribecontext=sj-main ; Added to try to fix wrong context on IB calls > qualify=yes > canreinvite=no > dtmfmode=rfc2833 > > When an inbound call comes in to sj0151, I get the following error: > > NOTICE[10777][C-]: chan_sip.c:26407 handle_request_invite: > Call from 'user1' (217.10.79.23:5060) to extension 'sj0151' rejected > because extension not found in context 'external-se'. > > Surely it should have looked in sj-main, not external-se? > > Also, the "Call from 'user1' is always 'user1' no matter which sipgate > account originated the call. The Callcentric numbers can't receive > inbound calls, the vohippo number shows "Call from 'user9'" as one would > expect. ALL of them look in context 'external-se', but the SJ > registrations should all be looking in 'sj-main'. What's more, it seems > to be struggling with pattern matching... The extension is passed > correctly (albeit to the wrong context, for 3 of the numbers), so the > following dialplan should pick them all up, surely?: > > [external-se] > ; Transfer any call from any SE external trunk to the IVR @ the office. > ; If the office is unavailable (no internet, for example), then go > to voicemail) > exten => _se.,1,Dial(IAX2/cloud/1000,30,r) > same => n,Voicemail(5000) > same => n,Hangup() > > However, it simply doesn't work. If I replace _se. with _se2489. (or > just se2489), it works fine (for calls arriving on the se2489 extension; > obviously the others bork). > > > Can anyone tell me what I'm doing wrong, based on the above? > > FWIW; this seems to have occurred because I've been attempting to prune > my dialplan; I used to have them all going into a single context, and I > picked them out & routed them individually. I am _trying_ to simplify > the structure/mess that is extensions.conf... but as a result I ran into > this little conundrum. The main problem is to resolve the "wrong > context"; I have a suspicion I could fix the "can't find extension" > problem by getting rid of the letters & using a purely numeric extension. > > Many thanks, > > Ade. I think your problem is that the "/se2489" or "/sj0151" in your register statement (for example) is not used to select or match the inbound SIP peer. When the call comes in from sipgate, it probably doesn't have a fromuser. The fromuser can be used to select the peer based on matching the [string] that names the peer. Otherwise, when Asterisk is looking for a matching peer section, I believe it only matches on the host from which the call comes. So when your call comes in from sipgate.co.uk, that is the only piece of information it uses, and so it always matches the first one, irrespective of the registration that originated the call. So I think you really do need to have a single peer section for all sipgate calls, pointing to one sipgate context in your dialplan that contains all your various extensions like se2489, sj0151, etc. Cheers Tony --
[asterisk-users] SIP trunks going to the wrong context
Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes limitonpeers=yes callcounter=yes vmexten=5199 nat=no ; SE registrations register => user1:passwo...@sipgate.co.uk:5060/se2489 register => user2:passwo...@sipgate.co.uk:5060/se1268 register => user3:passwo...@sipgate.co.uk:5060/se0845 register => user4:passwo...@callcentric.com:5060/se1777 register => user5:passwo...@sipgate.co.uk:5060/se4130 register => user9:passwo...@sip.vohippo.com:5060/se1413 ; SJ registrations register => user6:passwo...@sipgate.co.uk:5060/sj0151 register => user7:passwo...@callcentric.com:5060/sj1777 register => user8:passwo...@sipgate.co.uk:5060/sj0203 I then have a selection of #included files. The first defines se2489: [se2489] type=friend insecure=port,invite secret=password1 defaultuser=user1 fromuser=user1 fromdomain=sipgate.co.uk host=sipgate.co.uk port=5060 outboundproxy=proxy.live.sipgate.co.uk disallow=all allow=ulaw context=external-se qualify=yes canreinvite=no dtmfmode=rfc2833 The second defines sj0151: [sj0151] type=friend insecure=port,invite secret=password6 defaultuser=user6 fromuser=user6 fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk disallow=all allow=ulaw context=sj-main regcontext=sj-main ; Added to try to fix wrong context on IB calls subscribecontext=sj-main ; Added to try to fix wrong context on IB calls qualify=yes canreinvite=no dtmfmode=rfc2833 When an inbound call comes in to sj0151, I get the following error: NOTICE[10777][C-]: chan_sip.c:26407 handle_request_invite: Call from 'user1' (217.10.79.23:5060) to extension 'sj0151' rejected because extension not found in context 'external-se'. Surely it should have looked in sj-main, not external-se? Also, the "Call from 'user1' is always 'user1' no matter which sipgate account originated the call. The Callcentric numbers can't receive inbound calls, the vohippo number shows "Call from 'user9'" as one would expect. ALL of them look in context 'external-se', but the SJ registrations should all be looking in 'sj-main'. What's more, it seems to be struggling with pattern matching... The extension is passed correctly (albeit to the wrong context, for 3 of the numbers), so the following dialplan should pick them all up, surely?: [external-se] ; Transfer any call from any SE external trunk to the IVR @ the office. ; If the office is unavailable (no internet, for example), then go to voicemail) exten => _se.,1,Dial(IAX2/cloud/1000,30,r) same => n,Voicemail(5000) same => n,Hangup() However, it simply doesn't work. If I replace _se. with _se2489. (or just se2489), it works fine (for calls arriving on the se2489 extension; obviously the others bork). Can anyone tell me what I'm doing wrong, based on the above? FWIW; this seems to have occurred because I've been attempting to prune my dialplan; I used to have them all going into a single context, and I picked them out & routed them individually. I am _trying_ to simplify the structure/mess that is extensions.conf... but as a result I ran into this little conundrum. The main problem is to resolve the "wrong context"; I have a suspicion I could fix the "can't find extension" problem by getting rid of the letters & using a purely numeric extension. Many thanks, Ade. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
