similar to: Connecting Asterisks via SIP

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

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

> 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 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

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

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

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 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

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

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

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

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 =============>

I seem to be having problems with my 2 asterisk systems.. This works.. Call originating from the PSTN and then routed to the UA.. PSTN-->[pbx1]--IAX-->[pbx2]--SIP-->UA This doesn't work.. Call originating from UA to PSTN.. UA--SIP-->[pbx2]--IAX-->[pbx1]-->PSTN When the call is placed FROM the UA to the PSTN the call goes through and the PSTN phone rings but when it is

Ok.. I have done some more digging and the problem seems to be caused by chan_capi not detecting that the call has been answered.. I downgraded chan_capi from 0.2.3b to 0.2.2 and the system is working fine.. Kapjod, do you know of this problem? Later.. > I seem to be having problems with my 2 asterisk systems.. > > This works.. Call originating from the PSTN and then routed to the

Hi, Today's setup is : Legacy PBX1 with E1 ------- Leased line ----- Legacy PBX2 with E1 Prospective setup is : PBX1 ------- Asterisk GateWay1 (with Digium E1) ------ IP network ------ Asterisk GW2 (with Digium E1) ----- PBX2 Is there a way to tunnel, transport or translate Q.SIG signals between both PBXs ? The IP network is just used for point to point leased lines replacement :

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 =>