similar to: reverse the selection order of zap channels for outgoing calls

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

Hi I am using a 4FXO , TDM400P card I am able to call outside , after modifiying extensions.conf with exten => _9X.,1,Dial(ZAP/1/${EXTEN:1}) using this , I can only dial through one of the port , Actually I want to dial outside using round - robin search After reading the manuals , I have plans to modified the above line as exten =>

Hey all, I have a situation where I have 8 lines from the phone company in a hunt group coming in to my asterisk box. These are the same lines I'm using for outgoing calls ( named g0 ). The problem arises when someone dials our number at the same time asterisk tries to put a call out on one of the zap channels in the g0 group. This has happened twice that I know of so far, once to

I was playing around with the Flash Operator Panel, and came smack into a brick wall. We have a * box linked to a legacy Meridian System using a EuroIDSN link (TE405p) with 10 channels enabled. I also have several SIP extensions. What I wanted to do was to have a button for each of our (say) 32 users, 5 of which are on SIP. That leaves the other 27 on Zap. A potential of 27 users on 10

I have a problem that is probably so "doh" I will be embarrassed. However, I have spent all evening on this with no success: I have the following setup (asterisk cvshead as of today) 10 Channel EuroISDN<=>Asterisk<=>Meridian What I can do: Call from outside into the asterisk, dial an extension, and pass through to the meridian. WooHoo. What I can't do: Call from

On a TDM2400 with 3 FXO modules, is there a way to split each line into basically being its own trunk or another way to pull off the following scenerio: PBX has 12 inbound PSTN lines 1,3,5,7 are the 714 phone number hunt group 2,4,6,8 are the 888 phone number hunt group 9-12 are fax lines Customer wants outbound calls to go out in the following order: 8,7,6,5,4,3,2,1,12,10,11,9 Kerry

Hi, If I have a PRI with all channels grouped in group=1, I understand when I want to make an outgoing call that asterisk takes the first channel available. Is there any possiblity to "rotate" the channel taken? I was searching in Wiki but I could not find nothing about. Thanks, Alejandro

Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten =>

Hi all ! We currently have an asterisk box that is rather old (runs Asterisk 1.4.21.2), and it's connected to the PSTN with a sangoma A104d card. Now we have a new PRI at another location, and I use that occasion to build 2 new servers, one to replace our aging one and a new one for this new pri. So I downloaded the lastest libpri / asterisk / wanpipe driver, but the previous version of

Hi All, Does anyone have an example dial-plan and/or a way to use the queues to access outgoing calls on Zap channels when all the channels are congested? e.g. I have 20 users, who will be accessing Zap/g1 .. which has 3 channels I have another 5 users who will be accessing Zap/g2 .. which has only 1 channel. The outgoing numbers are dialled directly, and put through categorised extensions

Hi. We have 3 channel for _outgoing_ calls and would like to use them equally (by turn). Is there any 'queue' for outgoing calls? Now I have to select the second channel manually when the first is busy and so on. Does Asterisk have any functionality to do it automatically?

I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten => 1004,1,Dial(Zap/g1/NPANXXXXXX) I see the following on the asterisk console: -- Executing Dial("SIP/sbruton-b8ce", "Zap/g1/NPANXXXXXX") in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata.conf is as below:

Hi, I still cant dial out on Zap and I really have no clue why. I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31 channels each and able to receive incoming calls and fax perfectly. I've done this in my dial plan. exten => 111,1,Answer() exten => 111,n,Ringing() exten => 111,n,Wait(2) exten => 111,n,AbsoluteTimeout(30) exten =>

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

Hi, I have Installed * on a DL380 with a Junghanns 4BRI card and 0.0.2 driver. I have 3 BRI lines connected to SPAN(TE) 1,2,3 and 2 Cisco 7960 with SIP image. I am connected to french PSTN (France Telecom) whith Euroisdn signaling. I manage to call SIP to SIP, PSTN to SIP but not SIP to PSTN. Any idea? Thanks Gwenn Gael Marronnier Here is what I get and my configuration...

Greetings! I'm hoping someone can help me with what should be the most basic of problems. Essentially, I want to have certain calls on an Asterisk 1.2.25 (Yes I know its old, upgrade, etc... its on my roadmap) install go out a couple of analog lines and all other calls go out a PRI. The analog lines are setup in Zaptel group 1 and the PRI channels are in Zaptel group 0. Here is my relevant

Yes, I can make a call on that extension from other definity phone, if you mean it. -----Original Message----- From: Ken Godee [mailto:ken@perfect-image.com] Sent: 30 ÉÀÌÑ 2004 Ç. 19:14 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing Roman Bessyadovskii wrote: > Hi All. > > I connect

Hello Guys, I am trying to make Asterisk do dial-out calls. It doesn't even do test calls. It never calls. I tested everything and i am clueless. However i can call Asterisk and it picks up the phone and executes the dial-plan. However, my dial-plan is supposed to do outbound calls. Zap is configured correctly. I am using a TDM400 card from Digium with 4 Fxo ports and i have