similar to: Asterisk Java-Call Problem

Dear colleagues, I just have experienced a segmentation fault with Asterisk 13.5.0 and PJSIP 2.4.5. Both of them have been compiled on a standard Debian Wheezy 64 bit. I did not apply any patch or alter the sources of Asterisk or PJSIP in any way. Before compiling and installing, I removed all traces of all old Asterisk and PJSIP versions from my system very thoroughly. The segmentation fault

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

On Fri, 26 Jun 2015, Ludovic Gasc wrote: > 1. What's the "official" notation of each line: "=>" or "=" ? In the > wiki of Asterisk, I see very often "=>", however, what's the reason for > both syntaxes authorized ? Historical ? I'm not 'official,' but I have a strong preference for just '=.' Using

This is a followup to the posting earlier about Hunt Groups with Call Screening. I have implemented the following macro and for some reason the Macro does not exit and continue the context it was called from when the called party rejects the call - It always drops through to the NoOp at the end and connects the call. Below are two examples of the dial commands I am using to call the macro.

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

2015-06-26 17:11 GMT+02:00 Steve Edwards <asterisk.org at sedwards.com>: > On Fri, 26 Jun 2015, Ludovic Gasc wrote: > > 1. What's the "official" notation of each line: "=>" or "=" ? In the wiki >> of Asterisk, I see very often "=>", however, what's the reason for both >> syntaxes authorized ? Historical ? >>

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i

Trixbox 2.2 Beta with freePBX 2.2.0rc1 I have a setup that looks something like this in ASCII art: Teliax IAX Trunk ------+ | V Embarq PRI ----> Tandem switch ----> Ottawa Office Server------+ +--------------> Lima Office Server -----+|

Hello. I am running Asterisk v1.6 on Ubuntu 10.10 with FreePBX v2.8. When I call the voicemail for any of my extensions, the call just dies. On a softphone, I get no sound whatsoever; it just hangs up after a couple of seconds. On my handset attached to my SPA-3102, it get a sound like when you leave an analogue phone off the hook. I have three extensions setup and they all do the same thing.

I have a Asterisk server (PBX in a Flash) with Aastra 57i phones. When there is an incoming call the phone will display two buttons "answer" and "ignore". If you press "ignore" the call is dropped instead of sent to voice mail. The following is the log: -- Called 111 -- SIP/111-00001c14 is ringing -- Got SIP response 486 "Busy Here" back from

I have a little call recording script that I am running and it works fine, but I have one problem. I get CDR when a user calls into the extension, but I do not get CDR for the call that it makes outbound on # 17. Any idea why? Here is the extensions info: [default] exten => 2211,1,Answer exten => 2211,2,Wait(1) exten => 2211,3,Playback(/etc/asterisk/recording/getshop) exten =>

I did a NoOp and see what the callerid was and when coming from the SIP Ext->Voip it is set to the Extension Number of the SIP Extension (as you would expect). When coming from the Panasonic the CallerID is blank, I tried setting it to nothing again, and I tried setting it to the callerid of the voip provider, a sip extension id, the extension number on the Panasonic side, the zap channel

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

There was a post in the 'wiki' for an application to provide an outcall when there is a voicemail is left on asterisk. I am having a problem that this application will only work if the caller presses the pound sign at the end of recording. As most people just hang up, this application isn't working. Can any offer suggestions to accomplish this out call?

I have a simple cdr configured with the default tables, here is a row of a good cdr report calldate | clid | src | dst | dcontext | channel | ect ..... ect .... 2008-08-29 10:16:49 | "C. BOUTON" <40> | 40 | XXXXXXXXXXX | phonesystems | SIP/40-08776938 | ect ..... ect .... I have replaced the number by

Hi all, I have problem for receiving fax from multiple page fax that sent from fax machine (analog). The error is : WARNING T.30 Page did not end cleanly This is my dialplan [inboundfax] exten => s,1,NoOp(**** FAX RECEIVED from ${CALLERID(num)} ${STRFTIME(${EPOCH},,%c)} ****) exten => s,n,Set(FAXOPT(ecm)=yes) exten =>

I am trying to implement a Centralized Call Waiting System. I have red some document about asterisk group features to manage group and category of a sip channel. I have done a lot of test about it but always it doesn't work correctly if I transfer the call. This is the macro code I use for inbound calls. [macro-test] ; ${ARG1} - technology something like SIP ; ${ARG2} - resource.

Managed to implement this on asterisk v1.4.24.1, Also, Hangupcause updating to user field. However, this only works on the edge of my voice network (demarcation point) It does not work on my internal routing boxes as I use IAX to route between remote sites. I was thinking of using some sort of SIP variables to transport these results over the IAX trunk.. Any bright ideas folks???

Hello list, I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use. I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP

Hi Folks, I've been playing with call recording for our support department which was kinda going ok until I spotted something odd in the CDR. None of the support calls are being entered into the CDR properly. I'm using mysql as the back end and Areski's web based front end and all was going fine. The problem seems to be that the CDR doesn't get populated with the destination