similar to: False RING (incoming call) on Digium X101P FXO

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 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

hi, when i call asterisk (on x100p) i got this : CLI> -- Starting simple switch on 'Zap/7-1' Jul 13 15:03:34 ERROR[311316]: callerid.c:192 callerid_feed: fsk_serie made mylen < 0 (-9) Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4735 ss_thread: CallerID feed failed: Success Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4777 ss_thread: CallerID returned with error on channel

HI, im having problem with callerid. Im using tdm2400P and i get this from asterisk logs -- Starting simple switch on 'Zap/4-1' [Mar 24 02:07:48] ERROR[2358]: callerid.c:564 callerid_feed: fsk_serie made mylen < 0 (-1) [Mar 24 02:07:48] WARNING[2358]: chan_zap.c:6416 ss_thread: CallerID feed failed: Success [Mar 24 02:07:48] WARNING[2358]: chan_zap.c:6516 ss_thread: CallerID

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

Hi all, I've been trying to wrap my mind around this one for several days now. How can I 'debug' the CallerID reception on a Zap/POTS channel? I have a POTS line with CallerID and a Digium TDM11B card right now. I have my signalling set to ks for both sides, can make and receive calls just fine. But I never get CallerID on incoming calls. I get the following messages: Aug 11

[root@www root]# cat /proc/zaptel/1 Span 1: WCFXO/0 "Wildcard X101P Board 1" 1 WCFXO/0/0 FXSKS (In use) Asterisk CVS-HEAD-02/13/05-00:32:03, Copyright (C) 1999 - 2005 Digium. Feb 15 22:33:48 NOTICE[3002]: callerid.c:307 callerid_feed: Caller*ID failed checksum Feb 15 22:33:51 ERROR[3002]: callerid.c:261 callerid_feed: fsk_serie made mylen < 0 (-6) Feb 15 22:33:51

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: --

dear all, i am looking to enable CALLERID on an Asterisk system comprising a X101P FXO interface connecting to BT PSTN in the uk seems this is supported by the interface but there seems to be varying information on how to enable it in zapata.conf 1. usecallerid=uk 2. ukcallerid=yes being two of the configuration statements offered TIA GT

Hi all I?m a newbie to asterisk and I have install and configure asterisk 1.4.5 I have made some test and have face a strange behaviour I hava a simple dialplan when a call is receive from PTSN, [PSTN] exten => s,1,Answer() exten => s,2,Playback(intro-sicx) ; Listen to your voice exten => s,3,Dial(SIP/steph) exten => s,4,Hangup() I got the following when a call is

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

I tried implementing my * and it didn't pass the spouse factor at this time. I wanted to hook it up for outbound only at this point to get a better handle on the dial plans and the echo problem. I thought this might have been done before as a natural part of testing - but maybe not. In wcfxo.c I found this: if (!wc->offhook && !wc->ringdebounce) { if

Using the latest SVN of zaptel and asterisk, I can no longer receive incoming analog calls. The caller just hears it ringing but Asterisk doesn't pick up. I am seeing these error messages: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID

Hi, I installed a 1-port FXO on my Ubuntu 8.4. I was earlier only hearing a fast clicking sound and now I am not hearing any dial-tone. The FXO card has 2 slots: [phone | line ]. I hve connected the wall-phone-input to the "line" slot and "phone" to my home-phone. I do not hear any dial-tone on my home-phone. Asterisk seems to recognize my hardware....here are the relevant

The SM56 MODEM manual says it does. But when used with zaptel 1.2.12, nothing shows up. Yuan Liu

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 an FXO module in my TDM400P configured to receive caller*id (see zapata.conf below). I get a curious behavior: When I call this line with my cell phone, I see caller ID received just fine, with no warnings or errors.. When I call from another landline, I get different results: calling from external line, caller ID off: WARNING[1233021872]: chan_zap.c:4980 ss_thread: CallerID

I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten => s,1,Hangup (I'm just using the line for outbound dialing) -----Original Message----- From: asterisk-users-admin@lists.digium.com

When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time

X101P with Ambient md3200 chip on it, with the zaptel wcfxo driver.... Just an indication of how many people have got this to work would be useful. Cheers Mark.