similar to: Can't get x100p to answer the phone

I've followed the recent thread on caller id with UK British Telecom networks (where the caller id data is delivered before the first ring). My understanding is that if I use a recent CVS head (e.g. CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to do is include the line: usecallerid=uk In my zapata.conf (in the [channels] section) I've done this, but I get: Sep

we started using an ocfs file system about 4 months ago as the shared archived redo destination for the 4-node rac instances (HP dl380, msa1000, RH AS 2.1) . last night we are seeing some weird behavior, and my guess is the inode directory in the file system is getting corrupted. I've always had a bad feeling about OCFS not being very robust at handling constant file creation and deletion

(hardware in my computer: linux, asterisk, x100p, grandstream budge tone-100 ) Hi, When i run #asterisk ?v It show me a messages but when i try to incomming the call it show me that. Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'me@192.168.0.6' timed out, trying again Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

Hi, I've had this like since last saturday or something. I don't seem to be getting it to beta.centos.org tho, so i'll just make it public thru the channels i control. ftp://centos.upi.iki.fi/pub/centos/4.2beta/isos/sparc/ There is ISOs and .torrents Know yourself out and try it out please. If you have something less than Ultra Sparc, you're out of luck with this as it's

Hi, I've had this like since last saturday or something. I don't seem to be getting it to beta.centos.org tho, so i'll just make it public thru the channels i control. ftp://centos.upi.iki.fi/pub/centos/4.2beta/isos/sparc/ There is ISOs and .torrents Know yourself out and try it out please. If you have something less than Ultra Sparc, you're out of luck with this as it's

Hello, Can anyone help with the output shown below? It?s running on RH9, recent CVS of Asterisk and with one X100P card (2 channels), a budget tone 102 and Xlite softphone. CLI> -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no

Like some others on the list spandsp is segfaulting asterisk when recieving a fax. I'm on debian testing/unstable with freshly checked out asterisk CVS and sandsp. My libtiff version is 3.6.1. Here is the GDB output --- snip ----- Changed from phase 5 to 4 Start rx document - compression 2 Start rx page >>> CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase

Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to

I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. The system itself is happy and phone calls (between two parties) seem fine. Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file. To

Is anyone out there using app_dbodbc (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it? I was able to get it all working, but it causes * to segfault every now and then. It does not appear to be related to any specific function (ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I noload the module, everything works fine, but when its running, after calls to any of the

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

I'm having sporadic problems with my X100P card. When trying trying to call out I get a message on the console that the channel is busy even though I have bounced asterisk. It seems the card grabs the line and doesn't let go. When I try to call the number I get a busy signal. If I perform a rmmod wcfxo and restart asterisk everything retruns to normal. I have a TDP400P installed along with

Hi, I have got one X100P telephone card in my Asterisk server and it's working well. I have two phones lines and would like to install a second card so I can use both lines. I have installed the card and tried to set it up, but all to no avail ! Could someone outline the changes that I need to make (and in which .conf files) in order to get the second card going ? Thanks in advance,

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the

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

Specs: Oracle 9.2.0.4 OS is Redhat AS2.1 ocfs-2.4.9-e-summit-1.0.12-1 ocfs-tools-1.0.10-1 ocfs-support-1.0.10-1 ocfs-2.4.9-e-enterprise-1.0.12-1 Shared Storage: Dell/EMC CX600 naviagentcli-6.19.0.4.14-1.noarch.rpm PowerPath 4.4 My system was originally installed by Dell. Since then I've upgraded the OCFS and a few other pkgs. But ever since the beginning the ocfs drives mounted on boot.

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 Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik