similar to: Asterisk ---Toshiba

I have googled, and wiki'ed until blue. Is it possible to put T1---*----Toshiba CTX ? I have a TE405P, with one interface programmed for the T1, I am not sure how to program the 2nd port to mimick the T1 to the Toshiba. The Zapata.conf [channels] switchtype=national context=from-pstn signalling=pri_cpe usecallerid=asreceived echocancel=yes echocancelwhenbridged=no echotraining=400

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

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 +

Hy, I'm working with asterisk 1.2.4 and zaptel 1.2.4 With these version an the options in zapata.conf: answeronpolarityswitch=yes hanguponpolarityswitch=yes I don't detect polarityswitch. When asterisk reloads I see in CLI: May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring answeronpolarityswitch May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring

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

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

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

Greetings I was getting bad static crackle on a phone, so I reload from the latest CVS and did a make clean ; make install on zaptel, libpri and asterisk Now I get this error [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541 reload_config: Unable to get our IP address, Skinny disabled Urgent handler [chan_oss.so] => (OSS

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] --

Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird

i need your help regarding some issue related to the outband calls i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim with 2 ports when i try to call my phone number all time i receive message busy number this error just with g1. with g2 there is no problem i can call without issue can anyone see the CLI and tell me what is the problem thanks and regards == Parsing

Hello all, I have just installed a Wildcard X100P into an Asterisk box. I connected the line socket to the internal telephone system where I work. The card is identified to asterisk etc, however I am unable to recieve or make calls. When attempting to dial I get: Executing Dial("SIP/1106-ec8b", "Zap/1/644746") in new stack Called 1/644746 Zap/1-1 answered SIP/1106-ec8b Hungup

Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [channels] context=local usercallerid=yes hidecallerid=no

Hi, I recently installed a gentoo on a PC, purchased a TDM400 and installed asterisk. Kernel is 2.6.11-rc3 First, there seems to be a problem with my system: I must put the pci=routeirq to make the card detected. The modules are loaded ok (dmesg output) Zapata Telephony Interface Registered on major 196 ACPI: PCI interrupt 0000:00:08.0[A] -> GSI 16 (level, low) -> IRQ 16 Freshmaker

Hi, I'm trying to test my dial out function so I did something like this in extensions.conf exten => 999,1,Dial(Zap/g1/02601591) exten => 999,102,Congestion() My Zapata.conf looks something like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no ; PRI port 1 (E1) ; context=1 group=1 signalling=pri_cpe channel=>1-15,17-31 I am able to

Hi, i was able to fix this problem when i added the line pridialplan=local in the zapata.conf but it depends on your telco, i think. marcus -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Gesendet: Donnerstag, 6. April 2006 11:50 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Dial out on Zap Hi,

Hey Folks....I am on the 1.2 branch with the latest from Subversion. I've been having a rough go for the last several months integrating asterisk with out Altigen system. I can get calls inward just fine. I have zero missed interrupts on the digium 110p card. I have zero frame slips according to both sides. Outgoing calls sometimes work, but more often than not I get the following: --