[asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number

Tue Aug 18 01:26:19 CDT 2015

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> 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] *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] *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>
> 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/>
>
>
>
>
> --
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>
>
>
> --
>
> 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
>
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>

-- 
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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