similar to: no ring to callers?

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' --

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi, I have configured an Asterisk server with TDM01P (1FXO) for testing purpose. The interface I'm using is AMP. I want to configure my extension so that when I dial from my mobile phone to the asterisk line, I want it to transfer the call to any extension, say 3042 and after a particular number of rings, transfer the call to voice mail so that I can record my message. My Zaptel.conf is as

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

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hello All, I am trying to set up faxing using Asterisk@home 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing

I never heard anything on the AMP list, so figured maybe someone here might be able to help me sort this one out.. I was making some updates to my attendant config, which is really very basic, and now incoming call processing stopped. Not sure exactly what the heck happened, but figured maybe someone could help me with a clue as to what broke. Now incoming calls are not being answered at

Hi, using asterisk@home, with quadBri from junghanns.net I am facing a strange problem: I have set incoming routes for some extension / DID: [ext-did] include => ext-did-custom exten => 23,1,SetVar(FROM_DID=23) exten => 23,2,Goto(ext-local,23,1) exten => 57,1,SetVar(FROM_DID=57) exten => 57,2,Goto(ext-local,57,1) exten => 66,1,SetVar(FROM_DID=66) exten =>

Hello everyone, Ok. I am at a bit of a loss.... and would like someone to point me in the right direction...(btw www.google.co.za did not give me ANY solutions). The issue at hand is simple, I get asterisk (1.0.9) to answer the incoming call with no problems... it does the fax detection thing with app "Answer" and well it goes to the perfectly right context and sets the varibles

Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system

I don't succed in getting callerId on incoming calls on a zap trunk. I am using a zaphfc card When a call is received, one line in asterisk pbx says -- Executing SetCIDName("Zap/32-1", "") in new stack second parameters should be the caller ID, but it is not set The callerid is not hidden at source, so I think that is some kind of setting in zapata.conf I am using

Hello all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

Here is the output. These are Panasonic KX-TG2564's. Does something need to be set for the phones? I can call out fine, but all of the extensions seem to be busy. Starting simple switch on 'Zap/5-1' -- Executing Macro("Zap/5-1", "exten-vm|200@default|200") in new stack -- Executing SetVar("Zap/5-1", "FROMCONTEXT=exten-vm") in new stack

When I make a call out on the Faktortel number I am then able to transfer to call to my asterisk meetme room of 801 by hitting 'transfer' then '801' then 'send' on my grandstream phone. This connects my faktortel trunk (and who ever is on the other end) to my conference room I can then make another call using my local pstn service and set up a 3 way (or whatever number

Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Just did some testing.. root at vicky:~# cat /var/log/auth.log | grep "Set" May 30 21:41:05 vicky sshd[1568]: Set /proc/self/oom_adj from -17 to -17 May 30 21:41:07 vicky sshd[1574]: Set /proc/self/oom_adj to -17 root at vicky:~# ps faux | grep 1574 root 1574 0.0 0.0 70488 3404 ? Ss 21:41 0:00 \_ sshd: root at pts/1 root at vicky:~# ps faux | grep

Hi ALL Have installed asterisk@home 1.0 On FWD DID's, appears that 2 calls are generated to the inbound extention. I have confirmed this on a number of friends boxes also. Does anyone have a fix for this ? I set the DID simply to a custom context and it did the same... Anyone have a way to fix this ? Here is the output...... -- Accepting AUTHENTICATED call from 65.39.205.121, requested

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New

Running Asterisk Head 1.0.9. Below is a trace of a call delivered to my system which had no caller ID. For some reason, dialparties.agi shoves the incoming provider's IP address into the caller ID so you never have a call that is screened for PrivacyDirector. Is anyone else seeing this issue as well? Have I missed a patch? This call shows on the display with a name of "Unknown"