similar to: peers are using wrong contexts

Dear The Expert, I am very new with this, I have installed AsteriskNow, X-Lite as my SoftPhone, I am using SPA-3102. I have 3 extensions, me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (because the Linksys screen info said Registration Failed) Could any body please help? Many thanks in

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

I recently upgraded from asterisk 1.2.13 to 1.4.2 and am looking at using the users.conf file to setup my users, before i was using real time SIP which worked fine. However when i create a user in users.conf i am unable to register the user form a softphone, however that same softphone can still register a different the users i currently have setup form the sip.conf from real time. i've

Hello everyone, I have previously asked this question on the asterisk-video list, but I got directed here. I have a setup consisting of asterisknow beta4 (not sure if that is crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the local network. My computer has a USB-Camera installed, and now I would like to do some video calling with it, at least, so that the other user can

Hi, I am trying to setup a simple home voip service w/ * I have compiled and installed the svn source as a first step I am trying to configure SIP for inside my network. I have a handful of softphones and a few hardphones that I want to all be able to call each other I have configured users.conf with a single softphone(kphone) and have tried calling itself (ext 6000) and the demo from the

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

Hi, I've just built my first asterisk server. Current information: OS Version: Linux asterisk.visinet.com.au 2.6.18-8.1.8.el5 #1 SMP Tue Jul 10 06:50:22 EDT 2007 i686 i686 i386 GNU/Linux Asterisk Build: Asterisk 1.4.11 Asterisk GUI-version Revision: 1479 $ Server Date & TimeZone: Thu Sep 6 02:37:11 EST 2007 I've used the Asterisk GUI for setup with two IP

Can someone tell me how to setup a Aastra 75i phone? I have been trying to set it up and have pointed it to our asterisk server and selected http for download. What is the path? I have created two extension in asterisk for testing. I can't even get the phones to call each other. _________________________________________________________________ Lauren found her dream laptop. Find the PC that?s

I've got an issue where I am trying to route calls between Asterisk Servers. I can route calls inbound to a server but seem to have an authentication issue going out over the same sip account. It appears that my server isn't sending the second invite after proxy authentication request. I can't figure out why; any ideas would be greatly appreciated. Thanks! - Josh Here is my

i am configured asterisk-gui the "Queue Extension Configuration" but configure and register into queue.conf : [66666] fullname = Call Center strategy = ringall timeout = 5 wrapuptime = 5 autofill = yes autopause = no maxlen = 0 joinempty = no leavewhenempty = no reportholdtime = yes musicclass = default member => Agent/60010 member => Agent/60011 member => Agent/60014 but not

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

I have a SIP phone calling via a SIP trunk another asterisk system, that then sends the call out a ZAP channel. When I press any of the features defined in features.conf, The end user on the ZAP side hears the DTMF tones, and none of the features work. My DTMFmode on the SIP users definition is rfc2833 Asterisk console doesn't register that a feature is being recognized, any ideas? Below

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Hi, I have some problems and doubts connecting two asterisk servers. I have one asterisk (serverA), with 1 sip client registered (clientA). I have another asterisk (sever B), with another client (clientB). Now I want to call from client A to B and from B to A. Searching in google i find many configuration examples. For instance:

Hi, when I'm trying to call the following extension exten => 6002,1,Verbose(1|Extension 6002) exten => 6002,n,Dial(Agent/6002) exten => 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem