similar to: Not In Local Context

I have two contexts and there I have some sip clients and some iax clients, in the sip clients a have extentions like, 20, 21, 22, 23, etc; in the iax clients I have some extentions like 2000, 2001, 2002, 2003, etc. My extention is 2003 when I make a call the manager program show me that the extention 20 was originating the call (in red, and show my name/extention)! When I make a call to

Hello all, I'am trying to use 2 gnophones on my LAN. But I can't get any sound. Here is my configuration: 1. on PC1, I have: - Asterisk compiled from CVS (CVS-05/15/03) and it runs. - Gnophone binary version from Debian: 0.2.4+cvs.20020624-3 - a sound card working ( module es1371 for /dev/dsp) - I registered it as "alice" in extensions.conf 2. on PC2, I

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

I'm trying to create several contexts for extentions with different levels of access to features and I'm wondering how the heck do I include all the contexts so that you can call internal to any extention in another context without giving the features of the higher level context to the lower level context? ie..... [admin] include => local include => longdistance include =>

Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: "Registered to '65.39.205.121', who sees us as 68.14.203.254:4569" when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

Hi all, All of a sudden all my IAX DIDs have gone down. I couldn't find any reason other than that the ISP is blocking port 4569. DIDs register fine from my home server, but not from office server, which is not behind any NAT. SIP registers fine. I am trying to change IAX port but it apparantly IAX works only on 4569. Changing it in iax.conf doesn't do anything. Changing it is

I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called me@NuFone/1515480XXXX -- Call accepted by

So I'm trying to set up X-Lite for use with Asterisk, and I can't get it to work. I used a PDF I found on Voip-Info to set it up, and it's still not working, I'm using Nufone as the provider... I'll include my sip.cocnf and extentions.conf here. extentions.conf [CODE][outgoing] exten => _1NXXNXXXXXX,1,Dial,IAX2/scheda@NuFone/${EXTEN} [inbound] exten =>

I just loaded a new asterisk install (1.4.19) and found that the sip, iax, and extentions commands are missing from the cli and are not listed in help either? Any idea what could have happened or where these commands may have gone? Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

Hey, this is happening to anyone who I try this with. We get into the DISA, then hear the dial tone. Dial 1 then start dialing the number, and it hangs up. I thought adding a wait time after the DISA may help, I was wrong. Here is what I have thus far in the DISA extentions. [DISA] exten => 7,1,DISA(no-password||"Scheda" <565> 455-1337) exten => 7,2,Wait(45) exten =>

Hi I just signed up with voicepulse's voice connect service. then emailed me over configs for my extentions and iax i enter in all the info and when i start up * and do show registry it seems to be rejecting my login. Has anyone seen this before.. Any further insite will be greatly appreciated. thanks frankie (aim)cronparser (irc)crontibs 17006240093 -------------- next part

Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

Having some trouble with segfaults and sound quality all of a sudden (since I recompiled from the latest source) when 2 iaxComm clients connect. First off immediately after the server reports: <> <> -- Attempting native bridge of IAX2/4587@10.9.1.32:4569/1 and IAX2/4589/5 <> <>One or both client may sometimes segfault. Additionally, when they do get properly connected,

Dear Members, I am trying to make call from one Gnophone to another through the local Asterisk Server.All the three systems have local IP Addresses I created two users "sheeba" (extension 600) and "test" (extension 602) in iax.conf file: [sheeba] type=friend auth=plaintext host=dynamic secret=sheeba context=default allow=gsm permit=0.0.0.0/0.0.0.0

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

Most of the time, when someone calls in from the outside on a POTS line, and possibly over IAX as well, they don't hear any ringing sound while the internal SIP phones ring. If you call from an inside SIP phone, even forcing it into the incoming context, you hear the ringing. The outside calls can answer and talk fine; just no ring indication. Is there a setting that controls this?

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