Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone shed any light on this? Regards, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338
Steve Foy wrote:> Hi, > > I've got a fairly working Asterisk setup, with a few minor glitches, one of > which is very very irritating. > > Sometimes, during a call, the remote end just drops off. We're using software > SIP phones (SJPhone) connecting to * then out through analogue lines with > X100P cards. > > There is nothing in the logs and nothing on the console, the call just seems > to 'go away'! >Enable 'sip debug' at the CLI and send some detailed log file. /O
Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? Ciao, -b Quoting Steve Foy <steve@unite.net>:> Hi, > > I've got a fairly working Asterisk setup, with a few minor glitches, one of > which is very very irritating. > > Sometimes, during a call, the remote end just drops off. We're using > software > SIP phones (SJPhone) connecting to * then out through analogue lines with > X100P cards. > > There is nothing in the logs and nothing on the console, the call just seems > to 'go away'! > > Can anyone shed any light on this?---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean.
Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote:> Steve, > > this really is a FAQ. You need add to EACH (!) sip user something like > > disallow=all > allow=ulaw > allow=alaw > allow=gsmI do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338
Philipp von Klitzing wrote:> Hi! > >> I would add: >> reinvite=no in addition to canreinvite=no. >> It may do the trick. > > There is no such parameter as "reinvite=". Use "canreinvite=" only. > >> Ta >> SJ > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersWell... Googling... Few months ago produced that option. When was that option "dropped"?
On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote:> Steve, > > Did you ever figure out why this happens. I have had asterisk up and > running for a few weeks and all of a sudden this started happening.Exactly the same here, it was running fine for about a month or so. Then one day, a call disappeared, and gradually got more & more frequent. Nothing appears in logs or console. What phones are you using? -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338
Last 2 days I have noticed that more and more often calls are just being dropped. I can't find any logs or anything indicating that something is wrong. If I do a trace and wait for a call to drop I can only see hangup and nothing else. Sometimes calls do last for minutes without problem and sometimes they are dropped after about 30 seconds. Until yesterday it worked fine. I am using TE410P with 2 E1 connected trunks with h.323, sip and skinny phones on voip side. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Steve Foy Sent: Monday, February 09, 2004 3:35 PM To: Michael Nigrelli Cc: Asterisk-Users Subject: Re: [Asterisk-Users] Calls dropping off On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelli wrote:> Steve, > > Did you ever figure out why this happens. I have had asterisk up and > running for a few weeks and all of a sudden this started happening.Exactly the same here, it was running fine for about a month or so. Then one day, a call disappeared, and gradually got more & more frequent. Nothing appears in logs or console. What phones are you using? -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Might be, but even if you are not using voip, calls drop. I have a 2 E1 links and bridged calls between them drop from time to time. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ejay Hire Sent: Tuesday, February 10, 2004 5:46 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Calls dropping off I have this problem intermittently, and doing an asterisk -vvvvr showed "too many retries." hunting around with ethereal found a bad hub. -e> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On BehalfOf> Tomica Crnek > Sent: Tuesday, February 10, 2004 9:23 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Calls dropping off > > > Last 2 days I have noticed that more and more often callsare> just being > dropped. I can't find any logs or anything indicating that> something is > wrong. If I do a trace and wait for a call to drop I canonly> see hangup > and nothing else. Sometimes calls do last for minuteswithout problem> and sometimes they are dropped after about 30 seconds.Until yesterday> it worked fine. I am using TE410P with 2 E1 connectedtrunks> with h.323, > sip and skinny phones on voip side. > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On BehalfOf Steve Foy> Sent: Monday, February 09, 2004 3:35 PM > To: Michael Nigrelli > Cc: Asterisk-Users > Subject: Re: [Asterisk-Users] Calls dropping off > > On Mon, Feb 09, 2004 at 09:26:43AM -0500, Michael Nigrelliwrote:> > Steve, > > > > Did you ever figure out why this happens. I have had > asterisk up and > > running for a few weeks and all of a sudden this startedhappening.> > Exactly the same here, it was running fine for about amonth> or so. Then > one day, a call disappeared, and gradually got more & morefrequent.> > Nothing appears in logs or console. > > What phones are you using? > > -- > Steve Foy | http://www.unite.net > UNITE Solutions | Tel: 028 9077 7338 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users