<div dir="ltr">Glad to hear it's sorted.<br><br></div><div class="gmail_extra"><br><div class="gmail_quote">On 18 August 2015 at 17:08, Brendan Ord <span dir="ltr"><<a href="mailto:bord@staff.onthenet.com.au" target="_blank">bord@staff.onthenet.com.au</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Halt the wild goose chase ....<br>
<br>
<br>
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.<br>
<br>
Interestingly, it persisted through a "dialplan reload" and the equivalent of a "core reload" too ..<br>
<br>
[2015-08-18 17:04:30] VERBOSE[25543][C-00000000] app_dial.c: Called SIP/testing/0429920437<br>
[2015-08-18 17:04:30] VERBOSE[25543][C-00000000] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)<br>
<br>
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...<br>
<br>
Thanks for the help and apologies for the goose chase ..<br>
<span class="im HOEnZb"><br>
Brendan Ord<br>
OntheNet - Network Engineer<br>
P 07 5553 9222<br>
F 07 5593 3557<br>
</span><span class="im HOEnZb">Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)<br>
<a href="http://www.OntheNet.com.au" rel="noreferrer" target="_blank">www.OntheNet.com.au</a><br>
<br>
<br>
<br>
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-----Original Message-----<br>
</span><div class="HOEnZb"><div class="h5">From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Brendan Ord<br>
Sent: Tuesday, 18 August 2015 4:48 PM<br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number<br>
<br>
Hello,<br>
<br>
So, I found this line under macro-dialout-trunk, in extensions_additional.conf (FreePBX, so it controls the conf files mostly);<br>
<br>
exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})<br>
<br>
If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing..<br>
<br>
Here's a paste of a few things out of the two files that I thought were relevant to how FreePBX configured this trunk ...<br>
<br>
<a href="http://pastebin.com/5fRy2Ai9" rel="noreferrer" target="_blank">http://pastebin.com/5fRy2Ai9</a><br>
<br>
<br>
Brendan Ord<br>
OntheNet - Network Engineer<br>
P 07 5553 9222<br>
F 07 5593 3557<br>
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map) <a href="http://www.OntheNet.com.au" rel="noreferrer" target="_blank">www.OntheNet.com.au</a><br>
<br>
<br>
<br>
<br>
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<br>
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-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Bruce Ferrell<br>
Sent: Tuesday, 18 August 2015 4:38 PM<br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number<br>
<br>
just got back to my mail.<br>
<br>
What I'd do is go into /etc/asterisk and grep for OUT_3_SUFFIX in all the files<br>
<br>
once the file with that variable is located, we can figure out why it's adding it<br>
<br>
<br>
<br>
On 08/17/2015 11:26 PM, David Cunningham wrote:<br>
> Yes indeed.<br>
><br>
> Do you have the dialplan, eg from /etc/asterisk/extensions.conf?<br>
><br>
> Something is getting this OUT_3_SUFFIX variable and including it in a Dial to 172.22.4.12.<br>
><br>
><br>
> On 18 August 2015 at 16:21, Brendan Ord <<a href="mailto:bord@staff.onthenet.com.au">bord@staff.onthenet.com.au</a> <mailto:<a href="mailto:bord@staff.onthenet.com.au">bord@staff.onthenet.com.au</a>>> wrote:<br>
><br>
> Starting to make sense when I saw this line:<br>
><br>
><br>
><br>
> [2015-08-18 15:01:33] DEBUG[19366][C-00001cfc]: pbx.c:3785 ast_str_retrieve_variable: Result of 'OUT_3_SUFFIX' is '@CUBE'<br>
><br>
><br>
><br>
> But I can’t find where this is in configuration ..<br>
><br>
><br>
><br>
> Brendan Ord<br>
> OntheNet - Network Engineer<br>
> P 07 5553 9222<br>
> F 07 5593 3557<br>
> Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map <<a href="https://goo.gl/maps/p25WF" rel="noreferrer" target="_blank">https://goo.gl/maps/p25WF</a>>)<br>
> <a href="http://www.OntheNet.com.au" rel="noreferrer" target="_blank">www.OntheNet.com.au</a> <<a href="http://www.onthenet.com.au/" rel="noreferrer" target="_blank">http://www.onthenet.com.au/</a>><br>
><br>
><br>
><br>
> *From:*<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> <mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
> <mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>>] *On Behalf Of *Brendan Ord<br>
> *Sent:* Tuesday, 18 August 2015 3:44 PM<br>
><br>
><br>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion<br>
> *Subject:* Re: [asterisk-users] Asterisk 13 chan_sip trunk<br>
> appending @string to dialled number<br>
><br>
><br>
><br>
> David,<br>
><br>
><br>
><br>
> I should also note;<br>
><br>
><br>
><br>
> 246 is my extension, it has IP 172.22.3.238.<br>
><br>
><br>
><br>
> 172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway.<br>
><br>
><br>
><br>
> The trunk is called ‘testing’ at the moment. The route that selects this trunk uses a 9 prefix.<br>
><br>
><br>
><br>
> This system is in semi-production, so there might be fluff in the log from other active calls.<br>
><br>
><br>
><br>
> Brendan Ord<br>
> OntheNet - Network Engineer<br>
> P 07 5553 9222<br>
> F 07 5593 3557<br>
> Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map <<a href="https://goo.gl/maps/p25WF" rel="noreferrer" target="_blank">https://goo.gl/maps/p25WF</a>>)<br>
> <a href="http://www.OntheNet.com.au" rel="noreferrer" target="_blank">www.OntheNet.com.au</a> <<a href="http://www.onthenet.com.au/" rel="noreferrer" target="_blank">http://www.onthenet.com.au/</a>><br>
><br>
><br>
><br>
> *From:*<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> <mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
> <mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>>] *On Behalf Of *David Cunningham<br>
> *Sent:* Tuesday, 18 August 2015 2:39 PM<br>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion<br>
> *Subject:* Re: [asterisk-users] Asterisk 13 chan_sip trunk<br>
> appending @string to dialled number<br>
><br>
><br>
><br>
> Hi Brendan,<br>
><br>
> Can you attach an Asterisk log with "sip set debug on", "core set verbose 9" and "core set debug 9"?<br>
><br>
><br>
><br>
> On 18 August 2015 at 10:33, Brendan Ord <<a href="mailto:bord@staff.onthenet.com.au">bord@staff.onthenet.com.au</a> <mailto:<a href="mailto:bord@staff.onthenet.com.au">bord@staff.onthenet.com.au</a>>> wrote:<br>
><br>
> Hello,<br>
><br>
><br>
><br>
> 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,<br>
> something appends ‘@CUBE’ onto the end of the dialled number, as<br>
> per the following examples;<br>
><br>
><br>
><br>
> Asterisk log;<br>
><br>
> app_dial.c: Called SIP/test/0429123456@CUBE<br>
><br>
> chan_sip.c: Got SIP response 500 "Internal Server Error" back from<br>
> <a href="http://172.22.4.12:5060" rel="noreferrer" target="_blank">172.22.4.12:5060</a> <<a href="http://172.22.4.12:5060" rel="noreferrer" target="_blank">http://172.22.4.12:5060</a>><br>
><br>
><br>
><br>
> In the SIP SDP;<br>
><br>
> INVITE <a href="mailto:sip%3A0429920437%2540CUBE@172.22.4.12">sip:0429920437%40CUBE@172.22.4.12</a> <mailto:<a href="mailto:sip%253A0429920437%252540CUBE@172.22.4.12">sip%3A0429920437%2540CUBE@172.22.4.12</a>> SIP/2.0.<br>
><br>
> To: <<a href="mailto:sip%3A0429920437%2540CUBE@172.22.4.12">sip:0429920437%40CUBE@172.22.4.12</a> <mailto:<a href="mailto:sip%253A0429920437%252540CUBE@172.22.4.12">sip%3A0429920437%2540CUBE@172.22.4.12</a>>>.<br>
><br>
><br>
><br>
> 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<br>
> something different and the @CUBE persisted. I’m really not sure where this is coming from, and why.<br>
><br>
><br>
><br>
> Here is my trunk configuration;<br>
><br>
><br>
><br>
> PEER<br>
><br>
> type=friend<br>
><br>
> qualify=yes<br>
><br>
> nat=no<br>
><br>
> insecure=port,invite<br>
><br>
> host=172.22.4.12<br>
><br>
> dtmfmode=rfc2833<br>
><br>
> context=from-trunk<br>
><br>
> allow=ulaw<br>
><br>
> disallow=all<br>
><br>
><br>
><br>
> USER<br>
><br>
> type=friend<br>
><br>
> qualify=yes<br>
><br>
> nat=no<br>
><br>
> host=172.22.4.12<br>
><br>
> dtmfmode=rfc2833<br>
><br>
> allow=ulaw<br>
><br>
> disallow=all<br>
><br>
> canreinvite=no<br>
><br>
><br>
><br>
> Thanks for any help J<br>
><br>
><br>
><br>
> Brendan Ord<br>
> OntheNet - Network Engineer<br>
> P 07 5553 9222<br>
> F 07 5593 3557<br>
> Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map <<a href="https://goo.gl/maps/p25WF" rel="noreferrer" target="_blank">https://goo.gl/maps/p25WF</a>>)<br>
> <a href="http://www.OntheNet.com.au" rel="noreferrer" target="_blank">www.OntheNet.com.au</a> <<a href="http://www.onthenet.com.au/" rel="noreferrer" target="_blank">http://www.onthenet.com.au/</a>><br>
><br>
><br>
><br>
><br>
> --<br>
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> <a href="http://voisonics.com/" rel="noreferrer" target="_blank">http://voisonics.com/</a><br>
> USA: <a href="tel:%2B1%20213%20221%201092" value="+12132211092">+1 213 221 1092</a> <tel:%2B1%20213%20221%201092><br>
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> <tel:%2B61%20%280%29%202%208063%209019><br>
><br>
><br>
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> --<br>
> David Cunningham, Voisonics<br>
> <a href="http://voisonics.com/" rel="noreferrer" target="_blank">http://voisonics.com/</a><br>
> USA: <a href="tel:%2B1%20213%20221%201092" value="+12132211092">+1 213 221 1092</a><br>
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><br>
><br>
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