similar to: Wacky Failover Situation w/SIP - Bug?

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

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

It seems that I am having a heck of a time explaining my attempts at distributing ACD Queues to the list. Here's one little problem, that's only a piece of the puzzle. dundi.conf: 180q => global_dundi_q_pbx1,100,IAX,dundi1:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => global_dundi_q_pbx2,200,IAX,dundi2:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q =>

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

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

Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608)

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

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

> 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.

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

Hi Joseph, With iax servers dispersed across the internet, you could still use the below setup, it would work but it's not as secure as you would want it. I would then have a context for each server and use the IP address deny and permit statements. Also, you can have 1 server with a public IP and have the other servers behind a NAT register to the public server. There are really several

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? anybody know

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

Hi, Changing the line bellow helped to get incoming calls but I add to remove secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth required challenges). If someone could enable secret and still get incoming calls (in any SmartWare 5.X), please, do not hesitate to share here ... interface sip IF-ASTERISK bind context sip-gateway ASTERISK route call dest-table

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 confusion as

My asterisk environment is: ... -> [Asterisk PBX1] -> [Asterisk PBX2] -> [SIP Clients] Where the "..." are the normail landlines from where i am getting calls into my PBX1. As soon as i recieve a call into the PBX1 i use: exten=>BLAHBLAH,1,Dial(IAX2/PBX,10,tr) to forward it to the PBX2 but on the PBX2 side where i am supposed to choose between a number of sip clients, i

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

I have 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

I've added an important new application: app_transfer. This application is designed to allow Asterisk to request the transfer of an incoming call to a different extension. Consider the following diagram: Caller -> [ PBX1 ] -> SIP or IAX2 -> [PBX2] -> Transfer App A caller calls an extension on PBX1 which forwards to PBX2. PBX2 executes app_transfer, which requests that hte

Hello, Is it possible to transfer ss7 signaling over an FXO interface. I need to setup an ss7 test system composed by two Asterisk based IP-PBX systems with anlog interfaces only (FXO and FXS). I want to know if it is possible to connect the two IP-PBX as following: - FXS interface in PBX1 -----------------> connected to -----------------> FXO interface in PBX2 =============>