Glad to hear it's sorted.
On 18 August 2015 at 17:08, Brendan Ord <[email protected]> wrote: > Halt the wild goose chase .... > > > It was obviously something left over in the dial plan. Restarted Asterisk > seeing as though we're now after-hours and I can do interruptive work, and > it seems to have solved my @CUBE problem. > > Interestingly, it persisted through a "dialplan reload" and the equivalent > of a "core reload" too .. > > [2015-08-18 17:04:30] VERBOSE[25543][C-00000000] app_dial.c: Called > SIP/testing/0429920437 > [2015-08-18 17:04:30] VERBOSE[25543][C-00000000] app_dial.c: Everyone is > busy/congested at this time (1:0/0/1) > > This is expected, I need to review the dial-peer configurations on the > Cisco GW. At least it isn't throwing the suffix on the end anymore it > seems... > > Thanks for the help and apologies for the goose chase .. > > Brendan Ord > OntheNet - Network Engineer > P 07 5553 9222 > F 07 5593 3557 > Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map) > www.OntheNet.com.au > > > > > NOTICE: > > This e-mail and any attachments are private and confidential and may > contain privileged information. If you are not an authorised recipient, the > copying or distribution of this e-mail and any attachments is prohibited > and you must not read, print or act in reliance on this e-mail or > attachments. Any pricing information supplied via email is an estimate or > indicative only and may require a formal quotation to verify full terms and > conditions. > > > -----Original Message----- > From: [email protected] [mailto: > [email protected]] On Behalf Of Brendan Ord > Sent: Tuesday, 18 August 2015 4:48 PM > To: [email protected] > Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string > to dialled number > > Hello, > > So, I found this line under macro-dialout-trunk, in > extensions_additional.conf (FreePBX, so it controls the conf files mostly); > > exten => > s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}) > > If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing.. > > Here's a paste of a few things out of the two files that I thought were > relevant to how FreePBX configured this trunk ... > > http://pastebin.com/5fRy2Ai9 > > > Brendan Ord > OntheNet - Network Engineer > P 07 5553 9222 > F 07 5593 3557 > Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map) > www.OntheNet.com.au > > > > > NOTICE: > > This e-mail and any attachments are private and confidential and may > contain privileged information. If you are not an authorised recipient, the > copying or distribution of this e-mail and any attachments is prohibited > and you must not read, print or act in reliance on this e-mail or > attachments. Any pricing information supplied via email is an estimate or > indicative only and may require a formal quotation to verify full terms and > conditions. > > -----Original Message----- > From: [email protected] [mailto: > [email protected]] On Behalf Of Bruce Ferrell > Sent: Tuesday, 18 August 2015 4:38 PM > To: [email protected] > Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string > to dialled number > > just got back to my mail. > > What I'd do is go into /etc/asterisk and grep for OUT_3_SUFFIX in all the > files > > once the file with that variable is located, we can figure out why it's > adding it > > > > On 08/17/2015 11:26 PM, David Cunningham wrote: > > Yes indeed. > > > > Do you have the dialplan, eg from /etc/asterisk/extensions.conf? > > > > Something is getting this OUT_3_SUFFIX variable and including it in a > Dial to 172.22.4.12. > > > > > > On 18 August 2015 at 16:21, Brendan Ord <[email protected] > <mailto:[email protected]>> wrote: > > > > Starting to make sense when I saw this line: > > > > > > > > [2015-08-18 15:01:33] DEBUG[19366][C-00001cfc]: pbx.c:3785 > ast_str_retrieve_variable: Result of 'OUT_3_SUFFIX' is '@CUBE' > > > > > > > > But I can’t find where this is in configuration .. > > > > > > > > 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:*[email protected] <mailto: > [email protected]> [mailto: > [email protected] > > <mailto:[email protected]>] *On Behalf Of > *Brendan Ord > > *Sent:* Tuesday, 18 August 2015 3:44 PM > > > > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* Re: [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:*[email protected] <mailto: > [email protected]> [mailto: > [email protected] > > <mailto:[email protected]>] *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 <[email protected] > <mailto:[email protected]>> 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@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%[email protected] <mailto: > sip%3A0429920437%[email protected]> SIP/2.0. > > > > To: <sip:0429920437%[email protected] <mailto: > sip%3A0429920437%[email protected]>>. > > > > > > > > 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 < > 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 <tel:%2B1%20213%20221%201092> > > UK: +44 (0) 20 3298 1642 <tel:%2B44%20%280%29%2020%203298%201642> > > Australia: +61 (0) 2 8063 9019 > > <tel:%2B61%20%280%29%202%208063%209019> > > > > > > -- > > _____________________________________________________________________ > > -- 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 > > > > > > > -- > _____________________________________________________________________ > -- 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 > -- > _____________________________________________________________________ > -- 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 > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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
