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Asterisk is reporting the following error: [Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected ':', expecting $end; Input: : Always ^ here is the dialplan: exten => OUT,1,Gotoif($[$["${DB(AMPUSER/${ARG1}/recording)}" : "out=([^|]+)"] = Always]?r,1) exten =>

Hello, I have a TDM400 and currently have 2 of the ZAP Trunks configured on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4 with AMP version 1.10.010 In my Outbound Routing I have the Trunk Sequence set up so that 0 is Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is full, it does not open Trunk Sequence 1. I have found that this is true even if I

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

Hi i have installed AAH 2.6 and configured some extensions the calls are working fine. but if i dont answer a call then it says " the person at extension " and hangs up . it doesnt spell out the extesion number nor it goes to voice mail box. *************************** Asterisk CLI log **************************** dialparties.agi: Extension 200 is available...skipping checks --

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

Hello, I have setup a test server with asterisk/AMP and have several 7960's connected to it. The asterisk server has a public ip and all the 7960's are behind nat'd routers. When I try to call from extension to extension I get directed straight to voicemail. I do not have any cards installed and instead direct everything to an Ondo server. I have been told it's not an AMP

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

I have been shaking down a dialplan for SIP fax to efax. The basic senario is an ATA on the same subnet as the Asterisk 1.2 box (avoid RTP packet lose and thus fax crash), calling a 'fax extension' and envoking rxfax then email. I leverage off of context: from-internal-additional-custom, so as not to have it overwritten by FreePBX. In extension_custom.conf I have:

Skipped content of type multipart/alternative-------------- next part -------------- asterisk1*CLI> soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' --

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

I setup my aah box with a sip trunk at irisxa.iristel.net Incaming it is ok but when I try to dial 8 and the nr where I want to call I get all line is busy. In my log I have these: Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command' Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command' Feb 22 14:33:19 VERBOSE[2721] logger.c: --

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=

hi i've configured a TE205P on asterisk at home this is my zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone = it defaultzone = it and my zapata.conf signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes

I have an analogue trunk to an AT&T Definity. It has a DISA context defined. From a Definity handset call the analogue port extension 1008 and wait for dial tone from asterisk. It takes between 3&4 rings. Likewise from Asterisk SIP handset <PBX Access No><PBX Extn> takes nearly 10 secs to ring. Is this configurable? Ian Cowley -----Original Message----- From:

Nik, Just curious - what is your telco setup? Do you have PRI with the specified D channels? You need to make sure that your telco is set up to have the D channels on 16 and 47. When you first start Asterisk, or when you log on to the CLI, do you ever see messages stating the B channels are successfully started? Let us know. -MC -----Original Message----- From:

Hey all! I'm trying to do a bit of logic here so that a user only has to dial one code to pause/unpause in a queue (e.g. *0 will (un)pause depending on the users's state). My logic looks fine to me but every time ${PQMSTATUS} shows up empty. Here's the extensions.conf part.... exten => *0,1,NoOp(${PQMSTATUS}) exten => *0,n,Macro(user-callerid,SKIPTTL,) exten =>

Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing

Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.

Hi Folks, I'm have Asterisk version 1.2.1 with a A101 PRI card. I'm working with the CLEC to bring up the PRI and inbound calls are hanging up at his end after a few seconds. I ran PRI debug but it only gives me minimal insight. " Ext: 1 Cause: Unknown (16), class = Normal Event (1)" He ran a trace and the only difference he is seeing is a "ISDN interface explicitly