similar to: Calls hang in a conversation

Hi! We have some problems with calls that get disconnected in the middle of a call. We are using Asterisk 1.2.1 with a TE410P (2.gen firmware). When the call is disconnected Asterisk writes this to the log: Jan 9 14:56:17 DEBUG[4404] dsp.c: ast_dsp_busydetect detected busy, avgtone: 300, avgsilence 2090 Jan 9 14:56:17 DEBUG[4404] dsp.c: Requesting Hangup because the busy tone was detected on

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 =>

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),

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

Hi All I am using asterisk version cvs-v1-0-04/15/05 and spandsp 0.0.2pre18. I can receive and then email most faxes without issues, but recently I am having this issue when receiving faxes from a particular person. I can receive the faxes ok, but there are alot of bad rows as indicated by my logs below and the fax is not readable. I have included a good (from another user) and a bad fax. We

Hi All I am using asterisk version cvs-v1-0-04/15/05 and spandsp 0.0.2pre18. I can receive and then email most faxes without issues, but recently I am having this issue when receiving faxes from a particular person. I can receive the faxes ok, but there are alot of bad rows as indicated by my logs below and the fax is not readable. I have included a good (from another user) and a bad fax. We

I've been monitoring this problem for almost a month now. I realized that it is more extensive than what I described previously. I can very easily replicate this problem on every Zap channel. Following is the senario: 1. Call Zap/5 via say SIP/15 -> Zap/5-1 created and starts to ring 2. Call Zap/5 via say SIP/21 -> Zap/5-2 created and starts to ring 3. Hangup SIP/15 ->

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, We are running Asterisk 1.0.0 with a TE410P. Very often we exerience calls dropping in the middle of the call. I enable the full logging and saw a couple of suspicious messages right before the hangup. Thos could happen on a Zap-IAX2 bridge as well as on a Zap-Agent bridge... I see Nov 23 09:08:36 DEBUG[-1274020944]: Bridge stops because we're zombie or need a soft hangup:

I'm running * with a bog standard HFC ISDN card using zaphfc. Everything seems to work, including incoming calls, but I simply cannot make outgoing calls. This is very odd since the same card worked with the same configuration in another server. This is what I get from * debug. The only possible difference between the two servers that I can think of is that the HFC card is sharing an IRQ

Hello, I'm using asterisk-1.0.8 with BRI and spandsp-0.0.2_pre20. Modules app_txfax.so and app_rxfax.so are compiled and loaded sucessfully. It seems that the channel cant't detect the call as a fax-call 7612022801 is the calling faxmaschine 1209259 is my recieving fax extension logs are:

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

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,

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

Hi Guys, I have a really odd one here. We are dropping calls occasionally... there are no error messages being spat out, but I can see this suspicious behavior in the debug logs; Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is 'Other' Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is '(null)' Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is 's' Jun 30

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

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

Hey everybody, I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T. I've received several complaints about dropped calls. Reviewing the archives on PRI and dropped calls shows that I should set the resetinterval=never in the zapata.conf and restart. This hasn't helped. The dropped calls have to date only been on outbound calls. Usually within 2 to 3 minutes

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, I'm pulling my hair down and getting bold :-) ..... I have Asterisk between Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff Asterisk).... I'm trying to do just plain transfer of call from pbx to ISDN through Asterisk... It seems like PBX hangsup, when call is progressing with no apparent reason. I'd kindly ask for any advice or some working example for