similar to: PRI Moving channels?

Just throwing out this question. integrating with Altiware server. PRI appears to be okay. It keeps trying to move my call to a different channel...usually channel 1. This is the deal here: Moving call from channel 23 to channel 1 Then the following errors after no audio then hanging up manually: Mar 24 17:46:17 WARNING[1315]: chan_zap.c:7792 pri_fixup_principle: Call specified, but not

I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that

I would like to integrate * with an existing Altigen PBX. I want to spend as little money as possible to make it happen. My main goal is to inexpensively connect a branch office to the phone system. Eventually I would like to replace the Altigen system with an Asterisk server so I don't want to spend any money on Altigen hardware. Currently the Altigen has analog interfaces with a couple

Has anyone got Asterisk talking successfully to an Altigen PBX using h323? I can successfully make calls from Asterisk to Altigen, but calls from Altigen to Asterisk get a fast busy. There seems to be a lack of h323 example files (or maybe I'm looking in the wrong places) as well as a severe lack of h323 documentation from Altigen. Any pointers would be greatly appreciated.

System have been working great for weeks, using an average 40 of 120 dahdi channels. But today, I suddenly see scary things like this: -- Moving call from channel 5 to channel 7 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing

Greetings All- I have an * box with the NuFone H.323 channel driver installed. I also have an Altigen VoIP system with a PRI to the PSTN. I can sucessfully make a call from a SIP extension (snom190) to an H.323 extension (altigen phone) The thing I can't seem to make work is a call from a H.323 phone to a SIP extension. Here's the layout:

I'm attempting to do a first-time Asterisk install at home, firstly for use by my self and my family, and secondly as a learning experience. I've got a new house, and the previous owners removed all but one (1) phone jack. So I figured I might as well build a PBX. Functional goals include station-to-station calling, rudimentary auto attendant/voice mail, and perhaps tieing into the

Hi, I got a problem with PRI that I'm not sure how to solve. Asterisk sits between PABX and PRI. PRI is span 1 and PABX is span 2. After every single call (no matter in what direction) I get "pri_fixup_principle: Call specified, but not found?" and "pri_dchannel: Hangup on bad channel" messages and the channel in question is restarted. As far as I can see, all

Hi, in a PRI setup, the receiving side is changing the B channel at proceeding. It seems this sometimes breaks some logic (pri_fixup_principle) and then the hangup kind of breaks, release is not answered and a restart cycle is triggered (by remote side). Anyone can help me debug this ? I've seen many posts with simmilar issues but no answer/solution. This is happening on Asterisk 1.2.16 +

Hi Everyone. I have a client that uses an Altigen system. I am really new to PBX systems so all this is totally foreign to me. They currently have 5 inbound trunk lines and about 20 analog phones. >From what I can gather they are using the Altigen Quantum cards that support 8 extensions and 4 trunks. >From what I can gather the solution is a TDM04B and TDM01B to bring in the lines from

What am I doing wrong here? Why is this happening? libpri is version 1.0.7-1 (debian package) asterisk is version 1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2 -- Executing Dial("SIP/739-5935", "Zap/g1/0916000739") in new stack -- Called g1/0916000739 -- Channel 0/1, span 1 got hangup Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer:

Hello everybody again. I have a warning message in the CLI: *CLI> Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle: Call specified, but not found? *CLI> Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle: Call specified, but not found I don't know what it means. Can you help with this??? Thankyou very much. Bye bye... -------------- next part

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Hello list i have an issue with my dahdi_channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel => 17-31 context = default group = 63 but when i add in channel 1-15 like: channel => 1-15,17-31 i receive all

I've got a new setup (different building) where Asterisk is sitting between the PBX and phone company on a E&M T1 line. Mitel PBX <-> Asterisk <-> Phone company Inbound works. Asterisk gets the in-band digits from the phone company and hands the call off to the Mitel just fine. Outbound is weird. Asterisk seems to expect that the mitel will send routing information

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

ok guys I have a PHP GUI that will be great for both of you. direct editor to the whole file intact OR click to go to an extension. I will post a link to it tomorrow morning... as soon as I can get it off my production server HEHE.... it can do CRC checks on the *.cnf files and it will allow you to edit and parse out for you all your config entries with complex cnf files and default sample

After sometime I got my E1 PRI configured correctly in /etc/zaptel.conf and I now don't see any alarms on the E1, but I can't still dialout correctly, I enable every debug that I could though of and this is what I see: At the end the D-channel is down and I cannot even try to connect another call because it tell me that the zap channel is unavailable ,the other this that I notice is that I

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

Hello, I have a situation where a remote worker dials in to the asterisk server, enters the "secret code", then dials out via Disa on a PRI. This was all working great until this morning. Now the calls is placed out, connected but there is no voice from/to either phone. This is what shows on the CLI when the calls is bridged at a verbose of 4 and a debug of 1: [Jan 25 17:51:40] --