Walter Klomp
2004-Jul-29 07:43 UTC
[Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 alaw=1-124 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 signalling=pri_cpe channel => 1-15 channel => 17-31 ; Span 2 group=2 signalling=pri_cpe channel => 32-46 channel => 48-62 ; Span 3 group=3 signalling=pri_cpe channel => 63-77 channel => 79-93 ; Span 4 group=4 signalling=pri_cpe channel => 94-108 channel => 110-124
Robinson Tim-W10277
2004-Jul-29 08:43 UTC
[Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
This is quite common in some countries. Analogue lines are some times configured for 'calling party clearing', where an inbound call to an analogue line will hold the line for some minutes before timing out. Might this explain the behaviour? Rgds Tim -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Walter Klomp Sent: 29 July 2004 15:44 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 alaw=1-124 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 signalling=pri_cpe channel => 1-15 channel => 17-31 ; Span 2 group=2 signalling=pri_cpe channel => 32-46 channel => 48-62 ; Span 3 group=3 signalling=pri_cpe channel => 63-77 channel => 79-93 ; Span 4 group=4 signalling=pri_cpe channel => 94-108 channel => 110-124 _______________________________________________ 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
Peter Svensson
2004-Jul-29 09:17 UTC
[Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
On Thu, 29 Jul 2004, Walter Klomp wrote:> However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN > phone, the call does not get disconnected. My SIP phone goes quiet but > doesn't disconnect. If I a few seconds later pick up the PSTN phone again, > the connection is still there. Only if I hangup the SIP phone, the call gets > destroyed. It seems that Zap doesn't see the remote hangup...Normally a hangup at the b-subscriber (the receiving end) does not tear down the call immediatly, at least not for analog lines from the incumbent operator here in Sweden. I think it is something like 10-30s until the call is released in that case. Did you call an analog phone and how long did you leave it on hook?> Am I missing something ? How do I debug the Zap channels ?You need to set up debugging in the corresponding conf file for Asterisk. Debugging of the PRI signalling is then set up with pri debug span ??? or pri intense debug span ??? Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: <petersv@psv.nu> ! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF ------------------------------------------------------------------------ Remember, Luke, your source will be with you... always...
Soren Rathje
2004-Jul-29 09:54 UTC
[Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Walter Klomp wrote:> Hi >[snip]> > However, if I dial-in from the SIP phone to my PSTN and then hang up > my PSTN phone, the call does not get disconnected. My SIP phone goes > quiet but doesn't disconnect. If I a few seconds later pick up the > PSTN phone again, the connection is still there. Only if I hangup the > SIP phone, the call gets destroyed. It seems that Zap doesn't see the > remote hangup... >[snip] If memory serves me well (moved back to DK a year ago) then this is normal Singtel behaviour for subscriber-to-subscriber calling (it's so you can hang-up and go to another room, pick-up and continue). How long time before you see a hangup if you leave the PSTN side on-hook after the call ?? -- Soren
Peter Corlett
2004-Jul-30 01:55 UTC
[Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Walter Klomp <walter@aglow.com.sg> wrote: [...]> However, if I dial-in from the SIP phone to my PSTN and then hang up > my PSTN phone, the call does not get disconnected.This is normal and expected behaviour, at least for POTS lines I've used. When you receive a call on a POTS line, you can't clear it by just hanging up. On a POTS line from BT, you can force-clear an inbound call by hitting recall/hookflash then hanging up at the dialtone. The phone will ring for a few moments and then clear the call. --> IIRC the USA blew up their international telephone exchange very early in > the war.Was that bomb sponsored by AT&T or Cisco? - Mark Clayton and Tim Clark showing cynicism is alive and well in uk.telecom
Hi, Is there a way to get variable as DIALEDTIME or DATETIME ... with GET VARIABLE ? All my test always return unset variable. Else i can pass all variable need by args, but i would prefer more logic way to do it. Regards, -- Arnaud Pignard (apignard@frontier.fr) Frontier Online - Op?rateur Internet
Sergio Serrano
2004-Aug-29 08:30 UTC
[Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Hi, in Spain that process is correct. If you setup a communication between a caller and a called, if called phone hangs, in caller side hear a silence, but is a correct process. It's is due to in the called side you can hangup a phone and pickup other phone without lost communication. Regards, srsergio -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de Walter Klomp Enviado el: jueves, 29 de julio de 2004 16:44 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 alaw=1-124 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 signalling=pri_cpe channel => 1-15 channel => 17-31 ; Span 2 group=2 signalling=pri_cpe channel => 32-46 channel => 48-62 ; Span 3 group=3 signalling=pri_cpe channel => 63-77 channel => 79-93 ; Span 4 group=4 signalling=pri_cpe channel => 94-108 channel => 110-124 _______________________________________________ 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