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 b...@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
Hey!
Have you already seen it? http://gricela.laigriega.es/not.php?3
ortei...@tiscali.it
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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:
Brenden,
check the context, from-trunk, in the dialplan. Thtat's where this is
being added
On 8/17/15 5:33 PM, Brendan Ord 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
I recreated the trunk, and still get @CUBE added to the end of it.
I’m confused, nowhere in any configuration does this word appear anymore. How
would Asterisk stick this on? I grepped the entire conf directory looking for
a mention of it, and nothing ..
Brendan Ord
OntheNet - Network
Hi Bruce,
At the risk of sounding dumb ☺ And, realising that I mustn’t know how context’s
work properly (I guess they aren’t like Calling Search Spaces in Cisco-land).
I tried changing the context to from-internal and from-pstn with no change to
@CUBE being appended.
The from-trunk context
Hi Murthy,
You probably want [69.59.234.67] first so that it matches on inbound calls.
For outbound the question is what exactly do you specify in your dial? It
should be something like number@vonage-out.
On 14 August 2015 at 02:25, Murthy Gandikota murth...@hotmail.com wrote:
Hi All
On Mon, Aug 17, 2015 at 2:01 AM, Stefan Viljoen viljo...@verishare.co.za
wrote:
Hi List
Regarding this Asterisk instance as discussed previously (Asterisk
1.8.11.0)
that was consuming enormous amounts of file descriptors (100 000+ for about
50 simultaneous calls) it appears I have managed
Hi Kelvin,
Can you copy and paste in the dialog so we can see the call being set up
and the re-invite?
What version of Asterisk is it?
On 28 July 2015 at 20:05, Kelvin Chua kel...@gmail.com wrote:
I don't know if this is something asterisk can do at the moment but on my
setup, it does not.
Hi David,
http://pastebin.com/R4bsnmX7
I’ll start going through this as well and see if I can see anything.
Thanks for your help,
Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227
(Maphttps://goo.gl/maps/p25WF)
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
Hi
If we have a shared RealTime database for sip registration of multiple Asterisk
servers, is there a way to realize which Asterisk server registered sip phones ?
RegardsM.Shirazi
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Hi List
Regarding this Asterisk instance as discussed previously (Asterisk 1.8.11.0)
that was consuming enormous amounts of file descriptors (100 000+ for about
50 simultaneous calls) it appears I have managed to solve my problem by
upgrading the 1.8.11.0 Asterisk instance to an 1.8.32.3 Asterisk
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