similar to: Line drops

Hello to all, I have a strange problem with my asterisk. Line drops while i am in a call and without a reason.The line drops no matter if it is a incoming or outgoing call and it happen while i am talking on the phone (no silence detection problem). I tried to debug the situation and the only strange thing is the "got event On hook" (i guess..). I am thinking that it is a problem

Hi, I'm currently running CVS-HEAD 2005-09-03 I do plan to upgrade to the newest version, but need to do some testing with it first. In the mean time... does anyone know what these messages below are about? I've never seen it before, but when it happened it locked Asterisk up pretty good. Feb 10 10:16:51 DEBUG[14917] chan_zap.c: Echo cancellation already on Feb 10 10:16:57

Hi All, I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the phone will just ring and ring, even if I answer the phone on the other end. Whats strange is that the * phone will continue to ring even after I've answered and (sometimes) hung up the dialed phone. If I make an extension to just directly dial out on ZAP/1, its almost the same behavior, it will continue to

Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50

Hi all, Have been experimenting with agents and queues instead of placing calls direct to a user's phone extension, but I've run into problems with calls to both the agent and the extension which creates a zombie and double records calls abandoned etc. We're using a unique queue for each agent (only a handful of users) to try and get some agent/queue information to see what the

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

I have a trixbox 2.2 and Nortel santral that are speak each other. I use digium TDM100M 2 fxs-2fxo. After I made yum update I had met with some problems when I want to make any call from extension of trixbox to extension of nortel. When I attend to log (/var/log/messages) I meet with these messages as you see below. When I try to make any call from trixbox extension the call seems established but

I tried to add a couple of SIP phones (polycom 601s) to my existing asterisk installation. I can successfully make a call from the SIP phone to any other phone (inside or outside), but I can not make any calls to a SIP phone. Attached are the pertinent parts of sip.conf and extensions.conf. The log starts off normal with: Mar 26 09:51:15 DEBUG[4885] chan_zap.c: DTMF digit: 2 on Zap/55-1 Mar

Hi, Trying to trace an asterisk hang on a production (it had to be didn't it) system. The last thing before it crashed was [Mar 16 12:32:42] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 4 [Mar 16 12:54:34] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 4 [Mar 16 12:54:35] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED

Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at

Hi all, i use asterisk 1.2.7 and i have a problem with callerid. i use sangoma a200 cards. one fxo one fxs module i have these scenario - bob calls adam, where bob calls into my asterisk and adam picks up "from" my asterisk - bob and adam are speaking to each other - meanwhile eve calls adam, adam hears a beep, and knows there is an other caller on line. - bob and adam stop seaking

Hello, Can somebody please assist me with my problem. Currently I am using a Asterisk@HOme version 2.4 with a TE406P digium card. One the E1 is connected to a telco switch via an ISDN. May issue is that may incoming calls in the zap channels gets disconnected or release after 1 ring. I really dont know what setting should I change to increase the timeout of the ring. I have even tried upgrading

I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air. I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for incoming, e.g. host=gw4.telasip.com insecure=very qualify=yes type=user context=from-pstn Then

Hi There. I have the following setup : Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24 My problem is as follows : If I set up a very simple extensions.conf. when I dial from a fax machine, it seems as if no fax is being recognised. If I answer the call, I can hear the fax machine beeping. extensions.conf :

I've updated to the latest CVS as of today (and rebuilt RedHat 9). My setup is as follows: Wildcard X101P - channel 1 TDM400P - channels 2-3 - fxs cards with fxo signalling, channels 4-5 - fxo cards with fxs signalling I got CallerID to work on channel 4 with an old CVS, despite the usual "Didn't finish CallerID spill" message. However, as soon as I insert the following

I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running A@H 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on

We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from the second attempt to transfer the call, after it has actually been disconnected. Nothing is

We have an asterisk box with the following configuration: - AMD Athlon XP 2400+ - 512 MB RAM - SUSE Linux 10.1 - a Digium card TDM400P with 3 FXO - another Digium card TDM400P with 4 FXS - asterisk 1.2.7.1 - zaptel 1.2.4 I already checked that those cards aren't sharing interrupts (by cat /proc/interrupts): 0: 14119786 XT-PIC timer 1: 10 XT-PIC i8042 2:

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

Hi, sorry for the long debug output below. I configured Asterisk with AMP to send the whole number including the extensions of the callers to the called party. Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but doesn't seem to work. 033811234451 is the call id i configured, and it seems to use them, but the caller will only see a 0338189040 instead of my