Brendan Ord
2015-Aug-18 00:33 UTC
[asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number
Hello, I'm having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends '@CUBE' onto the end of the dialled number, as per the following examples; Asterisk log; app_dial.c: Called SIP/test/0429123456 at CUBE chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060 In the SIP SDP; INVITE sip:0429920437%40CUBE at 172.22.4.12 SIP/2.0. To: <sip:0429920437%40CUBE at 172.22.4.12>. As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I'm really not sure where this is coming from, and why. Here is my trunk configuration; PEER type=friend qualify=yes nat=no insecure=port,invite host=172.22.4.12 dtmfmode=rfc2833 context=from-trunk allow=ulaw disallow=all USER type=friend qualify=yes nat=no host=172.22.4.12 dtmfmode=rfc2833 allow=ulaw disallow=all canreinvite=no Thanks for any help :) Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>) www.OntheNet.com.au<http://www.onthenet.com.au/> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150818/dd2c7ac7/attachment.html>
Bruce Ferrell
2015-Aug-18 00:37 UTC
[asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number
Brenden, check the context, from-trunk, in the dialplan. Thtat's where this is being added On 8/17/15 5:33 PM, Brendan Ord wrote:> > Hello, > > I?m having what seems like a weird issue connecting Asterisk 13 > (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I > try dialling out via this trunk, something appends ?@CUBE? onto the > end of the dialled number, as per the following examples; > > Asterisk log; > > app_dial.c: Called SIP/test/0429123456 at CUBE > > chan_sip.c: Got SIP response 500 "Internal Server Error" back from > 172.22.4.12:5060 > > In the SIP SDP; > > INVITE sip:0429920437%40CUBE at 172.22.4.12 SIP/2.0. > > To: <sip:0429920437%40CUBE at 172.22.4.12>. > > As you can see, the @CUBE carries over into the SIP URI as %40CUBE. > The FPBX trunk name and outbound route were called CUBE (afaik, purely > descriptive) but I changed them to something different and the @CUBE > persisted. I?m really not sure where this is coming from, and why. > > Here is my trunk configuration; > > PEER > > type=friend > > qualify=yes > > nat=no > > insecure=port,invite > > host=172.22.4.12 > > dtmfmode=rfc2833 > > context=from-trunk > > allow=ulaw > > disallow=all > > USER > > type=friend > > qualify=yes > > nat=no > > host=172.22.4.12 > > dtmfmode=rfc2833 > > allow=ulaw > > disallow=all > > canreinvite=no > > Thanks for any help J > > Brendan Ord > OntheNet - Network Engineer > P 07 5553 9222 > F 07 5593 3557 > Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map > <https://goo.gl/maps/p25WF>) > www.OntheNet.com.au <http://www.onthenet.com.au/> > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150817/7d99bedb/attachment.html>
Brendan Ord
2015-Aug-18 01:31 UTC
[asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number
Hi Bruce, At the risk of sounding dumb ? And, realising that I mustn?t know how context?s work properly (I guess they aren?t like Calling Search Spaces in Cisco-land). I tried changing the context to from-internal and from-pstn with no change to @CUBE being appended. The from-trunk context looks pretty long, and includes a heap of other contexts as well ? this is all default configuration in FreePBX. I assume they?re doing most of the context leg work for us already in their distro?s ? Is there something I can stick in an email which might give a hint to my problem, or somewhere I should be looking? I was looking in extensions.conf at the contexts defined in there, was I in the wrong place? Thanks in advance, Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>) www.OntheNet.com.au<http://www.onthenet.com.au/> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bruce Ferrell Sent: Tuesday, 18 August 2015 10:38 AM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number Brenden, check the context, from-trunk, in the dialplan. Thtat's where this is being added On 8/17/15 5:33 PM, Brendan Ord wrote: Hello, I?m having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends ?@CUBE? onto the end of the dialled number, as per the following examples; Asterisk log; app_dial.c: Called SIP/test/0429123456 at CUBE chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060 In the SIP SDP; INVITE sip:0429920437%40CUBE at 172.22.4.12 SIP/2.0. To: <sip:0429920437%40CUBE at 172.22.4.12><sip:0429920437%40CUBE at 172.22.4.12>. As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I?m really not sure where this is coming from, and why. Here is my trunk configuration; PEER type=friend qualify=yes nat=no insecure=port,invite host=172.22.4.12 dtmfmode=rfc2833 context=from-trunk allow=ulaw disallow=all USER type=friend qualify=yes nat=no host=172.22.4.12 dtmfmode=rfc2833 allow=ulaw disallow=all canreinvite=no Thanks for any help ? Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>) www.OntheNet.com.au<http://www.onthenet.com.au/> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150818/79a0268b/attachment.html>
David Cunningham
2015-Aug-18 04:39 UTC
[asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number
Hi Brendan, Can you attach an Asterisk log with "sip set debug on", "core set verbose 9" and "core set debug 9"? On 18 August 2015 at 10:33, Brendan Ord <bord at staff.onthenet.com.au> wrote:> Hello, > > > > I?m having what seems like a weird issue connecting Asterisk 13 (FreePBX > 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling > out via this trunk, something appends ?@CUBE? onto the end of the dialled > number, as per the following examples; > > > > Asterisk log; > > app_dial.c: Called SIP/test/0429123456 at CUBE > > chan_sip.c: Got SIP response 500 "Internal Server Error" back from > 172.22.4.12:5060 > > > > In the SIP SDP; > > INVITE sip:0429920437%40CUBE at 172.22.4.12 SIP/2.0. > > To: <sip:0429920437%40CUBE at 172.22.4.12>. > > > > As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The > FPBX trunk name and outbound route were called CUBE (afaik, purely > descriptive) but I changed them to something different and the @CUBE > persisted. I?m really not sure where this is coming from, and why. > > > > Here is my trunk configuration; > > > > PEER > > type=friend > > qualify=yes > > nat=no > > insecure=port,invite > > host=172.22.4.12 > > dtmfmode=rfc2833 > > context=from-trunk > > allow=ulaw > > disallow=all > > > > USER > > type=friend > > qualify=yes > > nat=no > > host=172.22.4.12 > > dtmfmode=rfc2833 > > allow=ulaw > > disallow=all > > canreinvite=no > > > > Thanks for any help J > > > > Brendan Ord > OntheNet - Network Engineer > P 07 5553 9222 > F 07 5593 3557 > Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map > <https://goo.gl/maps/p25WF>) > www.OntheNet.com.au <http://www.onthenet.com.au/> > > > > -- > _____________________________________________________________________ > -- 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 >-- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150818/54523446/attachment.html>
Brendan Ord
2015-Aug-18 05:41 UTC
[asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number
Hi David, http://pastebin.com/R4bsnmX7 I?ll start going through this as well and see if I can see anything. Thanks for your help, Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>) www.OntheNet.com.au<http://www.onthenet.com.au/> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David Cunningham Sent: Tuesday, 18 August 2015 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number Hi Brendan, Can you attach an Asterisk log with "sip set debug on", "core set verbose 9" and "core set debug 9"? On 18 August 2015 at 10:33, Brendan Ord <bord at staff.onthenet.com.au<mailto:bord at staff.onthenet.com.au>> wrote: Hello, I?m having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends ?@CUBE? onto the end of the dialled number, as per the following examples; Asterisk log; app_dial.c: Called SIP/test/0429123456 at CUBE chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060<http://172.22.4.12:5060> In the SIP SDP; INVITE sip:0429920437%40CUBE at 172.22.4.12<mailto:sip%3A0429920437%2540CUBE at 172.22.4.12> SIP/2.0. To: <sip:0429920437%40CUBE at 172.22.4.12<mailto:sip%3A0429920437%2540CUBE at 172.22.4.12>>. As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I?m really not sure where this is coming from, and why. Here is my trunk configuration; PEER type=friend qualify=yes nat=no insecure=port,invite host=172.22.4.12 dtmfmode=rfc2833 context=from-trunk allow=ulaw disallow=all USER type=friend qualify=yes nat=no host=172.22.4.12 dtmfmode=rfc2833 allow=ulaw disallow=all canreinvite=no Thanks for any help ? Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>) www.OntheNet.com.au<http://www.onthenet.com.au/> -- _____________________________________________________________________ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150818/dc0a49dd/attachment.html>
Brendan Ord
2015-Aug-18 05:44 UTC
[asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number
David, I should also note; 246 is my extension, it has IP 172.22.3.238. 172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway. The trunk is called ?testing? at the moment. The route that selects this trunk uses a 9 prefix. This system is in semi-production, so there might be fluff in the log from other active calls. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>) www.OntheNet.com.au<http://www.onthenet.com.au/> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David Cunningham Sent: Tuesday, 18 August 2015 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number Hi Brendan, Can you attach an Asterisk log with "sip set debug on", "core set verbose 9" and "core set debug 9"? On 18 August 2015 at 10:33, Brendan Ord <bord at staff.onthenet.com.au<mailto:bord at staff.onthenet.com.au>> wrote: Hello, I?m having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends ?@CUBE? onto the end of the dialled number, as per the following examples; Asterisk log; app_dial.c: Called SIP/test/0429123456 at CUBE chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060<http://172.22.4.12:5060> In the SIP SDP; INVITE sip:0429920437%40CUBE at 172.22.4.12<mailto:sip%3A0429920437%2540CUBE at 172.22.4.12> SIP/2.0. To: <sip:0429920437%40CUBE at 172.22.4.12<mailto:sip%3A0429920437%2540CUBE at 172.22.4.12>>. As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something different and the @CUBE persisted. I?m really not sure where this is coming from, and why. Here is my trunk configuration; PEER type=friend qualify=yes nat=no insecure=port,invite host=172.22.4.12 dtmfmode=rfc2833 context=from-trunk allow=ulaw disallow=all USER type=friend qualify=yes nat=no host=172.22.4.12 dtmfmode=rfc2833 allow=ulaw disallow=all canreinvite=no Thanks for any help ? Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map<https://goo.gl/maps/p25WF>) www.OntheNet.com.au<http://www.onthenet.com.au/> -- _____________________________________________________________________ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150818/1c169ee4/attachment.html>