search for: pbx2

Hi Guys, i spent some time to figure this out (since i love how dialplan is written) but i decided to ask for your help guys. i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in pbx2 extensions.conf: i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr) in pbx1, i have: exten => 8029,1,Macro(stdexten,8029) and in stdexten macro: exten => s,n,Goto(s-${DIALSTATUS},1) exten...

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybo...

Hi. It is propbably a really naive problem, but I really couldn't find answer how to connect two Astrisks via SIP. I managed to do it via IAX without any problem. But this is a test installation and I would like to connect them via SIP. So I have two computers: pbx1 - 10.1.3.207 pbx2 - 10.1.3.204 pbx1 handles extensions 1xx and pbx2 for extensions 2xx. I would like to call user from pbx2 to pbx1 via SIP (note, I can call users within one PBX). What should be the configuration? I tried serveral configurations based on http://www.voip-info.org/wiki-Asterisk+-+dual+servers (esp...

Hi Guys, i am using the following config in pbx1: register => pbx1:endopass at 172.16.200.175 <pbx1%3Aendopass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=password context=[default] deny=0.0.0.0/0.0.0.0 permit=172.16.200.175/255.255.255.128 in pbx2: register => pbx2:endopass at 172.16.200.176 <pbx2%3Aendopass at 172.16.200.176> [pbx1] type=friend host=dynamic trunk=yes sercret=password context=[de...

Hi Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=0000000c, dsts=00000101) May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=00000101, dsts=0000000c) May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5....

I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a call from pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say that pbx2 will look for a [pbx1_outbound] .... oh dear... this doesn&...

I have a weird situation. A polycom phone is configured to use system pbx1 as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three systems have identical sip.conf files. The phone is registered on pbx1. I shut down the Asterisk application on pbx1. I make a call. The phone sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The phone sends an ACK followed by the INVITE with the credentia...

Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: register => pbx1:pass at 172.16.200.175 <pbx1%3Apass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=pass context=[default] ; i used the biggest context to avoid co...

Hi, I am trying to understand why some of my call transfers fail. My scenario is as follows: Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2 Step1: PBX1 extension 101 calls PBX2 extension 102 Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension 103 Step3: PBX1 extension 103 answers the call and transfers it to PBX2 extension 104 Step3 fails and extension 103 is reconnected to 101. Why is Step3 failing and h...

Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,NoOp(${CHANNEL}) ; Leg A exten => _X.,2,NoOp(${CHANNEL}) ; Leg B The problem is: after answering, [connect-jack] both priorities are executed, and right after executing them call drops. Log: -- Execu...

> Can someone tell me how Asterisk handles music-on-hold between servers? > Documentation for this is non-existent. > > Lets say user A, who is registered on pbx1, calls user B, who is registered on pbx2. > > 1. User A puts user B on hold. The moh that is played to user B should be specified according to user A. Which pbx box should this be set on? pbx1? pbx2? Both? > > 2. Is the situation any different if the 'trunk' between pbx1 and pbx2 is SIP or IAX? I'm using IAX trunk...

...ied it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new stack -- Executing [s at macro-myconnector:2] Read("SIP/pbx2-000004b2", "RESPONSE,beep,1,s,3,5") in new stack -- Accepting a maximum of 1 digits. -- <SIP/pbx2-000004b2> Playing 'beep.gsm' (language 'en') ......

Dear List I have notice this kind of problem between my two * box. My scenario is : Iax GSM IaxClient----->PBX1------------>PBX2-->TDM today CVS Stable V1 I use as Client FireFly with IAX2/GSM and try to call my PBX1 this server call PBX2 to terminate the call trought a TDM line (TE410P) but after PBX2 join the two call i can see the log below from my PBX1, i can speak for few second and after the FireFly hangup....

Marcelo Terres wrote: > I noticed another different behaviour. > > In older versions, when I call rasterisk, I receive some informations > about it. Fox example: > > [root at pbx2 ~]# rasterisk > Asterisk 11.22.0, Copyright (C) 1999 - 2013 Digium, Inc. and others. > Created by Mark Spencer<markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' > for details. > This is free software, with components license...

...ve 2 Asterisk servers connected via IAX2 connections. PBX1 is on the internet with a public IP Address - with PRI PBX 2 is behind a NAT router with IAX2 Ports forwarded 1-way audio is an issue with incoming and outgoing calls using the PRI. However whenever 1-way audio occurs, PBX2 can call PBX1 extensions and there are no issues. As well as a restart of asterisk on PBX2 clears up the problem. Any bugs in IAX2? Thanks aryn Aryn H. K. Nakaoka Tri-net Solutions 733 Bishop St. #170 Honolulu, HI 96813 http://www.trinet-hi.com <http://www.trinet-hi.com/&g...

Can someone tell me how Asterisk handles music-on-hold between servers? Documentation for this is non-existent. Lets say user A, who is registered on pbx1, calls user B, who is registered on pbx2. 1. User A puts user B on hold. The moh that is played to user B should be specified according to user A. Which pbx box should this be set on? pbx1? pbx2? Both? 2. Is the situation any different if the 'trunk' between pbx1 and pbx2 is SIP or IAX? Thanks, Doug.

...of dialplan: [cRETEUNICA] exten => testDTMF,1,Answer exten => testDTMF,n,Read(digito,,1) exten => testDTMF,n,SayDigits(${digito}) The point is that the recognition goes in timeout and i get an error on ast_waitfordigit_full -- Executing [testDTMF at cRETEUNICA:1] Answer("SIP/pbx2-000004ad", "") in new stack -- Executing [testDTMF at cRETEUNICA:2] Read("SIP/pbx2-000004ad", "digito,,1") in new stack [Jun 30 21:56:56] WARNING[5617]: channel.c:2558 ast_waitfordigit_full: Unexpected control subclass '-1' -- User entered nothing....

...lt on Centos 5.3 x86_64 dedicated server. out of the blue UDP stops responding .. and keep getting the following output: ---------------------------------- Opening message for the problem -------------------------------------------------- [Mar 21 09:57:04] ERROR[6748] netsock2.c: getaddrinfo("pbx2.server.com", "(null)", ...): Temporary failure in name resolution [Mar 21 09:57:04] WARNING[6748] acl.c: Unable to lookup 'pbx2.server.com' ----------------------------------------------------------------THEN ------------------------------------------------------------- [Mar...

...t;> >> it seems that on calls that i receive DTMF tones are handled correctly, >> but on calls generated from Asterisk to the world when the called side >> sends some DTMF digits they are not detected: >> >> -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") >> in new stack >> -- Executing [s at macro-myconnector:2] Read("SIP/pbx2-000004b2", >> "RESPONSE,beep,1,s,3,5") in new stack >> -- Accepting a maximum of 1 digits. >> -- <SIP/pbx2-000004b2> Playing ...

I have a weird situation. A polycom phone is configured to use system pbx1 as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three systems have identical sip.conf files. The phone is registered on pbx1. I shut down the Asterisk application on pbx1. I make a call. The phone sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The phone sends an ACK followed by the INVITE with the credentia...