I am experiencing a strange problem and wanted to know if someone has faced any similar issues or could provide me with a way to counter this problem. I am in the process of experimenting with asterisk and trying to setup a basic functional system. I have one TDM400P (single port) and one X100P. I am using one analog phone connected to the TDM400P and I also have a couple of Xlite SIP phones configured. I can make calls out to the PSTN and I can also receive calls. The problem happens when someone from the outside (PSTN) calls the Asterisk box. I have asterisk configured to forward the call to Zap/2 (analog phone). Zap/2 rings and I can talk to the person on the other end but if I hang up first then the other end does not see as the call being hung up. Asterisk CLI shows that Zap/1-1 (FXO) hungup but for some reason the other end thinks that the call is still up and does not disconnect unless the person hangs up himself. The confusing part is that if I initiate the call then this problem does not happen. Can someone tell me what is happening and how to resolve this issue? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040312/1110efb4/attachment.htm
Asterisk Learner wrote:> > > I am experiencing a strange problem and wanted to know if someone has > faced any similar issues or could provide me with a way to counter this > problem. I am in the process of experimenting with asterisk and trying > to setup a basic functional system. I have one TDM400P (single port) and > one X100P. I am using one analog phone connected to the TDM400P and I > also have a couple of Xlite SIP phones configured. I can make calls out > to the PSTN and I can also receive calls. > > > > The problem happens when someone from the outside (PSTN) calls the > Asterisk box. I have asterisk configured to forward the call to Zap/2 > (analog phone). Zap/2 rings and I can talk to the person on the other > end but if I hang up first then the other end does not see as the call > being hung up. Asterisk CLI shows that Zap/1-1 (FXO) hungup but for some > reason the other end thinks that the call is still up and does not > disconnect unless the person hangs up himself. The confusing part is > that if I initiate the call then this problem does not happen. > > > > Can someone tell me what is happening and how to resolve this issue? > > > > Thanks > > >On analog lines, the central ofice switch will signal that the remote end hung up the call, if I recall correctly, either by dropping loop current or using a brief polarity reversal. There may be other means as well. You need to make sure that you are using the correct settings to sense this properly. The reason you don't see this when you place the call is that, obviously, your end has no trouble determining when you have hung up, regardless of setting that affect the detection of the remote end hanging up. Stephen R. Besch
Hi! I have a strange problem with Asterisk (Asterisk 1.0.8-BRIstuffed-0.2.0-RC7k) on a VIA Samuel x86. When I make a call from the CLI either over IAX2 or SIP, my first call after the initial start of Asterisk works fine, even though upons starting Asterisk tells me "Read error on sound device: Resource temporarily unavailable". I hear the call over the loudspeaker. When I hang up (using CLI "hangup" command) and try to place another call, I get a "busy" signal over the loudspeaker and I get "Error reading from sound device (If you're running 'artsd' then kill it): Resource temporarily unavailable". If I issue a "hangup" again, I can dial out fine again. I don't have artsd running and nothing else besides Asterisk is using /dev/dsp (according to lsof). I have read somewhere that this might have to do with the sound card chip that I'm running (VIA Technologies, Inc. VT82C686 AC97 Audio Controller (rev 50)). Unfortunately I don't have the luxury of getting to see the debug output of /var/log/asterisk/debug on that machine, because it runs on a 128MB read only file system. I'm not loading chan_alsa.so, only chan_oss.so as I think this might have something to do with the problem. Any help would be great, or any hints into a possible direction. Thanks, Christoph
Hi All I am having a strange problem when I call from 1 RTC Client to another without Asterisk in between everything use to be fine but when asterisk is there as a Registrar a problem use to occur in many calls, Caller can hear the voice of the receiving side but the receiver cant be able to hear the caller for exact 8 seconds, conversation will become two way after 8 seconds but this problem is a big hurdle in proper establishment of a call Does anybody ever had this problem ? Any suggestions will be higly apreciated i have tried capturing packets but dont find anything abnormal Thanx in Advance __________________________________ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/