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Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

...half Of nik600 Sent: Monday, February 13, 2006 6:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] problem with outgoing calls Unable tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion) On 2/13/06, Michael Collins <mcollins@fcnetwork.biz> wrote: > 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...

Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over BackgroundDetect. If anybody used any of this command successfully, please help me. If possible, please let me

...half Of nik600 Sent: Wednesday, February 15, 2006 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion) On 2/14/06, Michael Collins <mcollins@fcnetwork.biz> wrote: > Nik, > > I'm not sure that "NOP" is correct, but I'm in the states so I'll to > defer to someone who knows E1/PRI. When I run zttool I have "OK" under > the alarms. Is there a way you can call the telco and confirm the > settings?...

...To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] problem with outgoing calls Unabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion) On 2/13/06, nik600 <nik600@gmail.com> wrote: > On 2/13/06, Michael Collins <mcollins@fcnetwork.biz> wrote: > > When Asterisk first starts up, it will attempt to "bring up" the B > > channels on any PRI circuits. If you are using A@H then you can log on > > to the Asterisk CLI (asterisk -r) and then do "stop now" to stop > > asterisk. Start up...

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

I'm having some trouble getting the channel status with an AGI script. #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->channel_status('Zap/1-1'); I am now stuck, and don't know how to get the return codes: -1 There is no channel that matches the given <channelname> 0 Channel is down and available 1 Channel

I've discovered that when dialing out using API's Originate action, a no answer is considered a failed attempt, while a busy is considered a successful attempt. The problem I'm having is that when I dial an invalid number, say a disconnected number that gives a fast busy, my CDRs are identical to those generated by a no answer attempt. Is there a way to distinguish between a no

Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

> you should mv the file (and in the same filesystem, so 'rename' is used) > You might want to chmod or even chown the file first as well. I wrote a little script that does all of this before the .call file is mv'd into the outgoing directory: cp /tmp/test3.call /tmp/test1.call chmod 666 /tmp/test1.call chgrp asterisk /tmp/test1.call chown asterisk /tmp/test1.call mv

Greetings all, This is my first post to this forum, so please be kind. I am looking to integrate Asterisk as our primary voice mail server (IVS) for our NEC NEAX 2000 IVS. We currently use a 6-port Mitel system. The two communicate via analog TDM ports and a MCI serial connection. Is anyone familiar with setting up a NEAX 2000 with Asterisk as an IVR server? I know I would need six FXO and

Question: I'm using a .call file to make some test calls. The call file works great. When I try the same thing with the manager 'originate' action I get something weird - the originate action looks for the 's' extension in my context, regardless of what I supply as the 'extension' argument. The .call file does what I expect - it finds exten _9.,1,Noop(Looks good).

This is slightly OT in that it isn't specifically *-related, but I was wondering what the members of the * user community felt about these two subjects. I've been perusing the OpenPBX.org mail list and the current hot topic is the fact that their project has come to a grinding halt. They are concerned that they don't have enough people working on their project. They feel that * has

...n Behalf Of nik600 Sent: Saturday, February 18, 2006 2:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] problem with outgoingcallsUnabletocreatechannelof type 'ZAP' (cause 34 -Circuit/channelcongestion) On 2/17/06, Michael Collins <mcollins@fcnetwork.biz> wrote: > Nik, > > This definitely helps! Please check your dial command. You've got > "Dial(Zap/0/mynumber)" and I think you might possibly want it to be > something like this: > Dial(Zap/1/mynumber) or > Dial(Zap/g0/mynumber) > > I don't reca...

Has anyone had experience with one or both of these cards? I'm in a position where I might need to recommend one over the other. I've read everything that I can find online, so now I'd like to hear of personal experiences. Everything I read on both cards is "5 stars! Awesome! It Rocks!" They both seem to have similar capabilities, similar pricing, etc. Could those of

The US has a Do Not Call list to which people can subscribe to prevent being called by advertisers. Federal laws (strengthened by some state and more local laws) assign penalties for calling people/phones on the DNCL. Is there a query gateway that Asterisk (or an app using Asterisk) can filter through to ensure a number is OK to call (not on the list) before calling it? -- (C) Matthew Rubenstein

I'm interested in knowing if anyone else has worked around this issue: I have an application that needs to check the status of the calls going through Asterisk about every 5 seconds or so. I don't want to do "asterisk -rx 'show channels verbose'" at the Linux command line 12 times per minute so I am looking at the AMI. I see that there isn't a manager command

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

Ok, here's a weird one. I have an AGI script where one user calls another. The call is answered. Everything is peachy. If the call is terminated by the CALLEE hanging up the call, then Asterisk returns control back to where the Dial() command left off, and I can check the return code of Dial(), ${DIALSTATUS} etc. That's all great. HOWEVER, if the CALLER hangs up the call, it seems as if

...Behalf Of nik600 Sent: Friday, February 17, 2006 12:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] problem with outgoing callsUnabletocreatechannelof type 'ZAP' (cause 34 - Circuit/channelcongestion) On 2/15/06, Michael Collins <mcollins@fcnetwork.biz> wrote: > Nik, > > Looks like you're making some progress. When I first started using A@H > I had trouble getting the outbound dialing to work. I wasn't sure where > to start, so what I did was skip the macros in the dial plan. I wanted > to play around with exac...