similar to: Can anyone shed some light on wht these calls were dropped?

Hi all, We keep getting these and all the calls between these two asterisk boxes get dropped. what is going on here, I have been trying to solve this problem on my own but maybe I don't have the trunk setup right. also I have posed the output of my full log of the machine with the zap interface, the other is using ztdummy. IAX.conf on machine 1: [general] port=5036 ;iaxcompat=yes

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

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

Problem: *Call* *waiting* comes in, I press flash to answer it, and the first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP. System: Analog stations and trunks running on a pair of TDM400's. It does NOT have * call* *waiting* from the phone company, and I have enabled it in all my conf files. The trunks are set to FXSKS and the stations are FXOKS. I am not using *call*

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

I have just finished implementing an Asterisk system for my place of business (first one), and after three days of flawless usage, Asterisk seems to have crashed. I wasn't running with '-g', so I don't have a core dump. Here's the sequence of events leading up to the crash: 1. call comes in on our TDM2400P 2. all of our phones (about 26 Polycoms) ring. (it's after

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

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

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),

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:

Hi evereybody, I'm trying to use musiconhold during dial tones. But I only can call earing dial tones instead of music. Now will see my configuration files. AGI File(using AGI script to EXEC DIAL) print "EXEC Dial Zap/g2/numberc||m\"; $res=checkresult(); Extension.conf exten =>_numberb,1,Answer exten =>_numberb,2,SetMusicOnHold,default exten =>_numberb,3,AGI,dial.agi

I have a Cisco 7970 that I am tying to get working. The phone powers up and registers but nothing else. I am using the development build. Any thoughts? The message I get when the phone powers on and registers. *CLI> skinny debug Skinny Debugging Enabled -- Starting Skinny session from 192.168.101.6 Recieved AlarmMessage Device SEP000F23FCA366 is attempting to register --

I noticed a problem this morning with our cdr logging. We have a cron job that places a call file into the spool directory having asterisk call itself to check to make sure its still handling incoming calls correctly, then queries the CDR database in mysql and makes sure that appropriate records exist. I can confirm that the call is happening correctly, but I'm missing records in the

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

On 1.2.x I have a random problem when a Zap/x channel flashes to transfer or make a three way call. The Zap/x-2 channel is created and the transfer or three way proceeds, but on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk goes crazy logging the problem. Here is an example debug log. This happens only once a day or so, with 100 or so users transfering and three

Hello, I have a TDM400P with 2x2 configuration of FXOs and FXSs. I set a test extension of '444' to dial out a specific zap trunk and call a local #. First time I call out to '444' everything works fine. If I hang up the call, and within 10 seconds dial the same number again, I get a fast busy. Seems it isn't letting go of the trunk or something, and I don't have a

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

Using tdm400. While transfering a call from outside to another extensions, while this "outside call" is waiting with music, the "another extension" call hangs up suddenly, and the call is back to the "outside call" suddenly. Wathcing logs: Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio while expecting 640 Jan 15 13:32:55 DEBUG[27850]

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come