I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP extensions ). I have upgraded libpri and zaptel to trunk, but I don't want to upgrade Asterisk to 1.2 until I've got this all sorted, one problem at a time! Here are my configs : /etc/zaptel.conf # Global data span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17 dchan=16 unused=18-31 loadzone = au defaultzone = au /etc/asterisk/zapata.conf [channels] context=from-onramp overlapdial=yes priindication = outofband switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe echocancelwhenbridged=yes echocancel=128 echotraining=800 rxgain=5 ; 0 txgain=-4.5 ; 0 busydetect=no pridialplan=local internationalprefix=0011 nationalprefix=0 usecallerid=yes hidecallerid=no callprogress=no group=0 channel => 1-15,17 /etc/asterisk/extensions: [from-onramp] ;exten => s,1,Playback(custom/aa_1) exten => s,1,Dial(SIP/116) exten => h,1,Hangup and here's some log info: asterisk*CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 asterisk*CLI> asterisk*CLI> asterisk*CLI> &&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&& -- Going to extension s|1 because of Complete received -- Executing Dial("Zap/1-1", "SIP/116") in new stack -- Called 116 -- Accepting call from '' to 's' on channel 0/1, span 1 -- SIP/116-5a95 is ringing &&&&&&&&&&&&&&&&& -- SIP/116-5a95 answered Zap/1-1 == Spawn extension (from-onramp, s, 1) exited non-zero on 'Zap/1-1' -- Executing Hangup("Zap/1-1", "") in new stack == Spawn extension (from-onramp, h, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' and going straight to a Playback command rather than SIP extension: asterisk*CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% -- Going to extension s|1 because of Complete received -- Executing Answer("Zap/2-1", "") in new stack -- Accepting call from '' to 's' on channel 0/2, span 1 == Spawn extension (from-onramp, s, 1) exited non-zero on 'Zap/2-1' -- Executing Hangup("Zap/2-1", "") in new stack == Spawn extension (from-onramp, h, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060228/23e18d9c/attachment.htm
Eric "ManxPower" Wieling
2006-Feb-28 20:55 UTC
[Asterisk-Users] incoming calls dropout on PRI over TE110p
Paul C wrote:> I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP extensions ). > > I have upgraded libpri and zaptel to trunk, but I don't want to upgrade Asterisk to 1.2 until I've got this all sorted, one problem at a time! > > Here are my configs : > > /etc/zaptel.conf > > # Global data > > span=1,1,0,ccs,hdb3,crc4 > bchan=1-15,17 > dchan=16 > unused=18-31 > loadzone = au > defaultzone = au > > /etc/asterisk/zapata.conf > > [channels] > context=from-onramp > > overlapdial=yesoverlapdial should usually be no in my experience.
broadbandvoice@comcast.net
2006-Apr-18 05:32 UTC
[Asterisk-Users] incoming calls dropout on PRI over TE110p
Try 1.2.3, it works fine. -------------- Original message -------------- From: "James Sturges" <thinking@1am.com.au>> I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of > the site. It is sending CRC errors )to Telsta, drops all calls once a day > for 1 second, calls getting stuck, quite unpleasant! > > I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri. > > James > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul C > Sent: Wednesday, 1 March 2006 4:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p > > > Paul C wrote: > >> I am running Asterisk 1.0.9 and have been running all my calls through a > >> VSP over a IAX2 trunk however we have recently purchased and connected a > >> TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make > > >> outgoing calls via it fine, however incoming calls are dropped after a > >> few seconds ( or as soon as a command like Playback, or the call is > >> picked up if forwarded to a SIP extensions ). > > >> SNIP << > > > > > overlapdial should usually be no in my experience. > > > Okay I've turned that to no with no change. I've just got off the phone to > Optus and apparently they had a client in melbourne last week and they fixed > > the problem by turning crc checking off at the optus end. I don't suppose > that was anybody on here ? > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060418/821adc3d/attachment.htm