similar to: Internode weirdness

Hi, Andrew. You are trying to solve two tasks: definition through what line the call came and a beautiful display of this information. 1. definition through what line the call came. If the username and password for inbound and outbound registration the same, then try the following: a) delete "register" lines. b) add option "callbackextension=Company1" to Company1 friend

Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but it does work. For prosperity, the SIP service is through Internode. Here is my "extensions.conf" file: exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one exten => s,2,Set(pseudodid=${SIP_HEADER(To)}) exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)}) exten =>

Hi Dmitriy and others and thanks for your help so far. The option "match_auth_username=yes" seems to have had no effect. From my reading, this option will try to match the username of the incoming SIP account to a section heading. If that is how it must work then i can see a big problem. I'm trying to present the receptionist with a nice display of which line the call came in on.

Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming don't work. I have sent this problem to this list a couple of times with little or no response, and I _really_ need some help to sort it out. I have an asterisk

This is one of the chronic problems. Try this option in sip.conf: match_auth_username=yes Carefully read the description, it is better to test in "after hours". 02.04.2015 2:50, Andrew Galdes ?????: > Hello all, > > I have an Asterisk server (Asterisk 10.12.4) with multiple sip > accounts with the same service provides. We have 8 phone numbers in > total. > >

Hello all, I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts with the same service provides. We have 8 phone numbers in total. Incoming calls from the public are all correctly directed to appropriate office handsets. However, the display on the reception phone (the only one i care about) is always showing the same "SIP/Account1_0843214321" rather than the account

I'm seeing this response to SIP calls originated in the following manner: Dial(SIP/${EXTEN}&SIP/{$DID},30,r) handle_response_invite: Re-invite to non-existing call leg on other UA. The response is from the second part of the dial. What exactly does it mean and how can I fix it? Thanks in advance Bruce

Jobst Schmalenbach <jobst at barrett.com.au> writes: > On Thu, Oct 05, 2017 at 02:57:18PM +1300, Clint Dilks (clintd at scms.waikato.ac.nz) wrote: >> On Thu, Oct 5, 2017 at 2:41 PM, Jobst Schmalenbach <jobst at barrett.com.au> >> wrote: >> [snip] >> Hi, >> >> Are you sure that your issue isn't related to the mirror that your >>

On Thu, Oct 05, 2017 at 02:57:18PM +1300, Clint Dilks (clintd at scms.waikato.ac.nz) wrote: > On Thu, Oct 5, 2017 at 2:41 PM, Jobst Schmalenbach <jobst at barrett.com.au> > wrote: > [snip] > Hi, > > Are you sure that your issue isn't related to the mirror that your > systems are selecting ? If they are using different mirrors I would try > using the

Hi, I''ve been trying to setup 2 DSL links in the office, both on the same firewall/router/gateway. the firewall is running kernel 2.6.11 (custom built on debian). this is the network diagram: 150.101.124.y/29 (internode''s gw) | | |150.101.124.x/29 (eth2)

The answer is attached. In summary it's not a bug in virt-df, nor df, nor the kernel. Rich. -- Richard Jones, Virtualization Group, Red Hat http://people.redhat.com/~rjones Read my programming and virtualization blog: http://rwmj.wordpress.com virt-df lists disk usage of guests without needing to install any software inside the virtual machine. Supports Linux and Windows.

On the Bombings Noam Chomsky The terrorist attacks were major atrocities. In scale they may not reach the level of many others, for example, Clinton's bombing of the Sudan with no credible pretext, destroying half its pharmaceutical supplies and killing

I recently (a few weeks ago) installed 1.9.18p8 in order to get better performance with an Access Database, which along with adding a TCP_NODELAY in the [global] section has that working like a charm. Thing that's bugging me now is smbstatus. When I run it I see open files listed when I know that they aren't. The following output shows quite a few files open that aren't really. The

On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:07:35 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > <snip> > >> Here

Totally understood. If you remain "on vacation" (we wish eh?!) and get further messages like that from the list s'ware, just accept my standing apologies and delete them. Very welcome re letter! Very best, Chris ------- Forwarded message follows ------- Send reply to: "Alana O'C" <alanaoc at internode.on.net> From: "Alana O'C"

Hello, In order to overcome a known performance bug in perl-5.8.8-10 in centos 5 (see https://bugzilla.redhat.com/show_bug.cgi?id=196836) I downloaded the perl package from fedora 8 (http://mirror.internode.on.net/pub/fedora/linux/releases/8/Fedora/source/SRPMS/perl-5.8.8-30.fc8.src.rpm) and mod_perl

https://bugzilla.mindrot.org/show_bug.cgi?id=1798 Daniel Black <daniel.subs at internode.on.net> changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |daniel.subs at internode.on.ne | |t -- You are receiving

Hi guys, sorry for the long e-mail, i'm only trying to give as much information as i think is relevant to my problem (console log, sip.conf and extension.conf parts). i've been practicing with callback for a while, but i'm at a dead end. I hope somebody can help me to move on. i have troubles getting two calls bridged together. Scenario is the following: - asterisk calls my

On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > Tested with X-Lite and it worked fiine. Is there some way to replace > "Anonymous" with a config parameter? > > Thanks for your kind help > > ---------------------------------------- > > From: murthy64 at hotmail.com > > To: asterisk-users at lists.digium.com >

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --