search for: pbx1

...exten => receive,n,System('/usr/local/bin/fax2mail -p -f "${FAXFILENOEXT}" --cid-number ${CALLERID(num)} --cid-name "${CALLERID(name)}" --dest-name "Sir/Madam"') a previous debugging showed: *- for a fax from myfax.com that was received successfully:* pbx1*CLI> > Channel 'DAHDI/1-1' fax session '53', [ 034.021683 ], channel sent 59 frames (1180 ms) of energy. pbx1*CLI> -- Channel 'DAHDI/1-1' fax session '53', [ 040.489601 ], STAT_EVT_HW_CLOSE st: WT_HW_CLS rt: WCLSNCLS pbx1*CLI>...

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...

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...

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/${EXTEN},20,tTWwr) in pbx1, i have: exten => 8029,1,Macro(stdexten,8029) and in stdexten macro: exten => s,n,Goto(s-${DIALST...

...ch a " * DTMF Shooting" the logfiles recognized this (see the channel types!) : -- NOTICES -- Nov 6 09:53:26 WARNING[22637] res_features.c: Bridge failed on channels mISDN/1-1 and Zap/1-1 Nov 6 10:05:28 WARNING[22902] res_features.c: Bridge failed on channels SIP/40-0815e778 and SIP/pbx1-08281bc8 Nov 6 10:15:38 WARNING[23393] res_features.c: Bridge failed on channels SIP/40-0826c530 and IAX2/pbx1-1 DTMF Tone Log : Nov 6 05:00:33 DTMF[18215] channel.c: SIP/50-0824f1e0 : A Nov 6 08:44:05 DTMF[21660] channel.c: Zap/1-1 : A Nov 6 09:43:00 DTMF[22520] channel.c: SIP/pbx1-08274fb8...

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%3Aendop...

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=[defau...

See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Thanks. -- James <--- SIP read from zzz.zzz.zzz.44:9363 ---> NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: "xxx-xxx-xxxx" <sip:xxxxxxxxxx at pbx1.mydomain.com>;tag=467525dd6fac949do0^M To: <sip:pbx1.mydomain.com>^M Call-ID: 707176dd-38f4779d at 192.168.1.140^M CSeq: 118907 NOTIFY^M Max-Forwar...

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+-...

Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1...

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 recon...

.... I would like to watch calls through the management interface, but I don't know how to identify which call an event belongs to or in other words how to associate a call and uniqueid field of event. Let's say I send the following manager command: action: originate channel: sip/12125551111@pbx1 callerid: 12125551111 MaxRetries: 1 WaitTime: 10 Application: AGI Data: callback.agi|2&12125551111&12125552222 Then I'm receiving the following events: Uniqueid: 1091642334.98 Event: Newchannel Callerid: State: Down Channel: SIP/pbx1-fc4f Uniqueid: 1091642334.98 Event: Newcallerid Ca...

> 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 pbx...

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...

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...

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...

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?...

Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable...

Hello, I'm trying to patch Asterisk for uses wth BroadVoice. I'm running Asterisk@Home. Here is the Error: [root@pbx1 asterisk]# patch < broadvoicesip2.txt can't find file to patch at input line 8 Perhaps you should have used the -p or --strip option? The text leading up to this was: -------------------------- |Index: channels/chan_sip.c |=================================================================== |...

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 => global_dundi_q_pbx3,300,IAX,dundi3:${SECRET}@${IPADDR}/${NUMBER},nopartial extensions.conf(PBX1): [global_dundi_q_pbx1] include => one_queue_a...