similar to: Registration failed though configured.

do not know why. Anybody has a clue what could be wrong ? Is this a bug ? [I rebooted asterisk, and now it works.] Regards Axelle. Logs of failed registration: > sip show users Username Secret Accountcode Def.Context ACL NAT IMSI208011234567890 sip-local No RFC3581 IMSI208302141472352 sip-external No

Hi list, I have searched through asterisk command lines but haven't found how to do this: - can I list the phones (callerid or IMSIs?) currently registered ? If I do "dialplan show" that lists the configuration I loaded, e.g [ Context 'sip-local' created by 'pbx_config' ] '2102' => 1. Macro(dialSIP|IMSI1) [pbx_config] '2103'

I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full:

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

Hi list, I have a user, referenced by his IMSI (IMSI208300618462231), who is assigned to extension 2111 in /etc/asterisk/extensions.conf and sip.conf (see below).

I have a simple problem with sialing a SIP device. I'm SURE it's a syntax problem, but I dunno what it might be. Here are the debug messages: == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) -- Executing Goto("Zap/1-1", "2111|1") in new stack -- Goto (default,2111,1) -- Executing Dial("Zap/1-1",

Hi all, I would like to add a new partition to my ZFS pool but it looks like it''s more stricky than expected. The layout of my disk is the following: - first partition for Windows. I want to keep it. (no formatting !) - second partition for OpenSolaris.This is where I have all the Solaris slices (c0d0s0 etc). I have a single ZFS pool. OpenSolaris boots on ZFS. - third partition: a FAT

Hi, I'm trying to automatically have the dialplan assign an extension to a roaming phone on my network. I tried the following without success: exten => 3001,1(readop),BackGround(beep) exten => 3001,n,Read(digito,vm-youhave,3) exten => 3001,n,SayDigits(${digito}) exten => 3001,n,Set(ROAM=${digito}) exten => 3001,n,Set(DB(roam/ext)=${digito}) exten =>

First off, please, for the love of God, don't cremate me, if I should already know the answer to this! I've installed a small setup for an office who wanted to be able to talk to each other instead of having to rely on MSN to communicate. Weird request, I know, but hey, we do what we need to do to get paid. I installed soft phones, gave everyone an extension, and bingo, they can call and

I have the following in extentions.conf: exten => 2111,1,Dial(SIP/2111 at gw1.langley) exten => 2111,2,Voicemail(u2111) exten => 2111,3,Hangup exten => 2111,100,Voicemail(b2111) exten => 2111,101,Hangup I have the following in sip.conf: ; Cisco 1750 [gw1.langley] type=friend host=172.16.17.1 context=default canreinvite=no Like the ATA, lots of stuff doesn't work on the 1750

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

We have quadBRI configured 1 port in TE mode 2,3,4 ports in NE mode. We are trying to place a call from the phone connected to BRI card port #4 to city number through ISDN line connected to port #1. Number successfully dialed. Person on the other end answering the line. But conversation can't last more then 10 seconds. Below is a log of such call. Its not clear for me why we appear in

Hi All, I am using Asterisk for one of my projects in OpenBTS. I am having the age old problem of "extension not found" when try to make a call from one registered SIP phone to other registered SIP phone (two mobile phones connected to Asterisk via OpenBTS). The exact error thrown on Asterisk CLI is *"chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to

Hi all, -------- I have installed a TDM400 with one active FXS port (TDM10B) an connected it to a Siemens Euroset 2015 analogue phone. I have installed some smom IP phones to the network as well and configured them as usual (sip.conf). For configuring the TDM10B I have used FXO signalling in /etc/zaptel.conf and in /etc/asterisk/zapata.conf. I definded the TDM channel and the Snom phones to the

I'm having problems with my dial plan. I have the following in my extentions.conf: [incoming] ; ; Incoming calls via the PSTN land here ; exten => s,1,Answer exten => s,2,DigitTimeout(5) exten => s,3,ResponseTimeout(10) exten => s,4,BackGround(dial-exten) exten => s,5,Wait(15) exten => s,6,Congestion exten => s,7,Wait(10) exten => s,8,Hangup ; ; Let incoming calls

libswfdec/swfdec_as_string.c | 31 +++--- libswfdec/swfdec_color_as.c | 9 - libswfdec/swfdec_interval.c | 2 test/image/.gitignore | 2 test/trace/Makefile.am | 27 +++++ test/trace/array2-8.swf |binary test/trace/array2-8.swf.trace | 40 ++++++++

Hi Dave, >> context=openbts >> callerid=4735202222 >I see you are using OpenBTS. To my understanding, OpenBTS does not >support caller ID, so I don't think it can work. >But as I have the same issue as you, I'd be glad to be wrong ! :D Let me know. Disregard my answer. I just tested the callerid on my OpenBTS and it worked. So the problem you encounter must be