similar to: No Caller ID from FXO Problem

I have TDM400P , with 1 FXS and 1FXO I'm tring to forward all incoming calls to a SIP phone in the context where all calls from the fxo come i have : exten => s,1,dial(SIP/phone1000,5) the phone rings but when i answer the sip phone ( phone1000 ) is connected but the phone from which i'm ringing still rings. Here is the log from asterisk : *CLI> -- Starting simple switch on

Is it possible to call a macro, which is defined in extensions.conf from a realtime extension configured in Mysql. Beacuse when i try i receive an error - no such context. -- Executing Macro("SIP/1007-2165", "dialnumber_wvm,1004,SIP/1004") Dec 11 12:51:04 WARNING[22551]: app_macro.c:100 macro_exec: No such context 'macro-dialnumber_wvm,1004,SIP/1004' for macro

11-10-2004 there was a subject: Re: Where did USE_SIP_MYSQL_FRIENDS go?: on asterisk.user list. >All db specific code has been removed from the code in favor of the >currently-in-development "RealTime" method of configuration from >database. >You are most likely not using the 1.0 stable branch. >You need to use the new RealTime configuration method. And currently,

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup

I have * box on machine with external ip address and internal one I'm tring to register to it from to machines - one from innternet ( everything is ok - in sip.conf nat=yes)\ and the other one is in the internal network (in sip.conf - nat=no ) and it say 403 Forbidden? Any Ideas ? here are the logs and configs From the external SIP Client whic registers.

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

I'm using asterisk with XLite everything is working. But in the asterisk console I always receive some notice of Registration failed . What is the reason for this? How Can be fixed? message : Jul 1 16:18:29 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'damian <sip:damian@10.1.1.11>' failed for '10.1.1.11' Asterisk and Sip phones are all in one

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

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 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

Aloha, I'm finishing up the final touches on this install, and have run into an odd problem. I can't seem to get Caller ID on the analog phone lines working. It's a Digium AEX 410 card. I have Verbose set and a line to print the CID: I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf and users.conf [analog] include=>default exten =>

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 =>

How can i control the codec for the calls. For example I have 3 SIP phones registered to asterisk The firs two are in the local area network (behind nat)- I want to use g711 between them and to connect directly (canreinvite=yes) and the third is in internet - want all calls to it and from it to use g729 and media to go through asterisk. So if Phone 1 calls Phone 2 the codec to be g711, but when

While on conversation with another party, I dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep'

I'm tring to compile fax applications on Debian system. the spandsp library compiles ok, and when i try to patch the make file in apps directory as is said in the instructions it returns errors. I'm using cvs version of asterisk . -------------------------- voipgw:/usr/src/asterisk/apps# patch < Makefile.patch patching file Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of

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

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 is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

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