similar to: Problem with incoming calls from FXO

No Caller ID comes from the FXO line ( The caller id is on and is working with a standard phone) in zapata.conf everything looks fine usecallerid=yes hidecallerid=no When the call comes in there are some warnings in Asterisk Console -- Starting simple switch on 'Zap/4-1' Jun 22 11:20:24 NOTICE[213006]: callerid.c:281 callerid_feed: Unknown IE 17 Jun 22 11:20:24 NOTICE[213006]:

I have an Asterisk server running todays CVS (updated it just in case that was the problem). It has 3 X100P cards. The first two lines I use as my normal work lines and the third is my fax line which I use with SpanDSP. I run Fedora Core 4. I have a problem that the third X100P does not answer the call. From the console I can see that there is an incoming call with the following: --

Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

I have a normal backup phone (and an alarm panel) sharing the POTS line with the Digium X101P FXO: | | Wall |>---+------X101P FXO as Zap/5 | | | Phone & Alarm Whenever the Phone is used, Asterisk sees a 'false ring' signal immediately when the phone is hung up. The Alarm panel dials out nightly at around 1AM, and each time it completes the call, Asterisk

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line

I am attempting to transfer a number that comes in on an FXO port back out the same port. The service has 3 way calling and transfer and these options are specified in zapata.conf . Some config... zapata.conf ------------ [channels] ; context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=yes callprogress=no

Hi... I have the next problem.. I have a FXO card with i can make calls but i cant recive calls. At the consol, i get the next error: -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel

Hi all. I am unable to answer calls coming into asterisk over PSTN. (UK) I want to have a call answered by my TDM400P/FXO module and forwarded to a sip phone. When I make a call from the PSTN to the BT line installed on my FXO module the sip phone rings however, when i pick up the call using the sip phone, the incoming call is not answered/routed by asterisk. As a result the sip phone is left

Howdy, 1 x TDM400P card with 1 x fxo module. 1 x BT Pots line. Location - UK Calls work fine outbound but i'm unable to pickup the inbound calls. Asterisk debug: Asterisk -vvvvvvvvvvcg *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Answer("Zap/1-1", "") in new stack

Calling from PSTN let extension 601 ring twice, hang up and starts over again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SIP/102,20) exten => s,3,Voicemail(u102) exten =>

HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>

Hi. I'm experiencing that bug with flash on zaptel. That's the problem: Zap/A call Zap/B Zap/B flash transfers to Zap/C Now Zap/A is online with Zap/C Till now all ok... but now if Zap/C wants to transfer again, it can't... the debug says that it got a WinkFlash when call not up or ringing (as attached below, Zap/10 is Zap/C in my example) Apr 2 09:14:01 DEBUG[32789]: File

Hi, I am a beginner in asterisk. Implementing it in my dept in India using TDM400b card with asterisk, zaptel, libpri version latest of CVS HEAD Callerid on my system is coming tough. Asterisk doesnot finishes the callerid spill and Cancells it. After going through code in Callerid.c and chan_zap.c I found that my line is providing caller id of length 8867. Flow enters in zt_call and

Starting - oh - three weeks ago I started getting this when I compiled zaptel stuff: In file included from /lib/modules/2.4.26smp/build/include/linux/spinlock.h:6, from /lib/modules/2.4.26smp/build/include/linux/module.h:11, from wct4xxp.c:31: /lib/modules/2.4.26smp/build/include/asm/system.h: In function `__set_64bit_var':

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line