[asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number
Brendan Ord
bord at staff.onthenet.com.au
Tue Aug 18 02:08:25 CDT 2015
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: 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> --
> 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
More information about the asterisk-users
mailing list