[asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number
David Cunningham
dcunningham at voisonics.com
Tue Aug 18 06:36:08 CDT 2015
Glad to hear it's sorted.
On 18 August 2015 at 17:08, Brendan Ord <bord at staff.onthenet.com.au> 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
>
>
>
>
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>
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>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] On Behalf Of Brendan Ord
> Sent: Tuesday, 18 August 2015 4:48 PM
> To: asterisk-users at lists.digium.com
> 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: 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 4:38 PM
> To: asterisk-users at lists.digium.com
> 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 <bord at staff.onthenet.com.au
> <mailto:bord at staff.onthenet.com.au>> 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:*asterisk-users-bounces at lists.digium.com <mailto:
> asterisk-users-bounces at lists.digium.com> [mailto:
> asterisk-users-bounces at lists.digium.com
> > <mailto:asterisk-users-bounces at lists.digium.com>] *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:*asterisk-users-bounces at lists.digium.com <mailto:
> asterisk-users-bounces at lists.digium.com> [mailto:
> 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 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> --
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> >
> >
> >
> >
> > --
> >
> > 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
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> --
> _____________________________________________________________________
> -- 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
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> 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
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