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David Cunningham

2015-Aug-18 06:26 UTC

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

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/>
>
>
>
>
> --
> _____________________________________________________________________
> -- 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
>


-- 
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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Bruce Ferrell

2015-Aug-18 06:37 UTC

head link

[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
>
>

Brendan Ord

2015-Aug-18 06:48 UTC

head link

[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

?

??
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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 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

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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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