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Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. When I pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if I dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

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

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

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

Hi, Some calls I make trough my PSTN asterisk gateway just hangup after some minutes. Even if I'm using sip or iax. I have callprogress=no busydetect=no in my zapata.conf. Anyone help? Or tell me what to look at /var/log/asterisk/debug. I didn't find anything wrong. [endpoint]---iax or sip----[asterisk]----E&M----PSTN. As endpoint I had tested another asterisk box (with a FXS),

Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network. For some reason, CDR is billing time even though the "busy tone" was detected. It's also logging the call as ANSWERED. Is this normal behavior? Seems a little odd to me. I have this as the first 3 lines of my zapata.conf [channels] busydetect=1 busycount=3 CVS HEAD updated late

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work fine. To avoid problems with the phone line (german E1) I called the provider, he did a 45 min. route

Hi Everybody, I am trying to use SIP (Sipura 2000) to connect to Asterisk which then dials out a local number using the Digium E100P. We have purchased the G729 codec licenses from Digium and loaded them into Asterisk successfully. However, the call drops immediately after being answered with the debug error message saying something like: "channel.c:2646 ast_channel_bridge: Didn't get a

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

Hi all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty'

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, I'm trying to install a Junghanns quadbri for a few days but i stay with an asterisk error. (Everyone is busy/congested ) Asterisk is working with a Fritz PCbut from one year and now i want to add the quadbri. The quadbri card has been configured in NT mode and with no 100 ohms S/T termoination. (I'm not sure if the S/T parameter is correct) I have installed the bristuff package

Hi, For a significant time now (since about 0.2.0-rc8n) the qozap driver has become very verbose if an ISDN line is not connected... I get the messages below every couple of seconds in the asterisk logs. The "flaw" in the messages is the "Alarm cleared" message - The alarm cannot possibly be cleared because there is no physical media connected into that port!!! (BTW - All

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. I'm getting the following error at random intervals on my TE410P with Asterisk CVS-03/30/04-11:49:01-CEST. I have two spans active, one connected to my Telco, the other to a Siemens PABX. Both spans display this behavior at random intervals. All calls are dropped when this happens. Spans are not necessarily in use when this happens.

I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,