Michael Sampson
2005-Dec-19 07:29 UTC
[Asterisk-Users] Can't call out on ZAP channel - need help
I'm trying to connect to another PBX via an T-1 interface. I have a T100P card. On the CLI I get the error "Everyone is busy/congested at this time (1:0/0/1)" When I try to dial out of the T-1 line from an SIP softphone. I have posted this question a few times here and at the asterisk forum, but can't get anyone to respond. I've seen other people on forums with the same problem but no one has ever given much of a solution. Does someone at least know what the next step in debugging this problem would be. In the file /var/log/asterisk/full I get the error "Unable to create channel of type 'ZAP'" Here are my configs. Zapata.conf ------------------ ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn ;signalling=fxs_ks signalling=pri_net ; pri_cpe= PRI slave ; pri_net = PRI master switchtype=qsig pridialplan=local resetinterval=never ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf channel => 1-23 ------------------------- Zaptel.conf ------------------------- # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0" # channel 1, WCT1, unhandled for now # channel 2, WCT1, unhandled for now # channel 3, WCT1, unhandled for now # channel 4, WCT1, unhandled for now # channel 5, WCT1, unhandled for now # channel 6, WCT1, unhandled for now # channel 7, WCT1, unhandled for now # channel 8, WCT1, unhandled for now # channel 9, WCT1, unhandled for now # channel 10, WCT1, unhandled for now # channel 11, WCT1, unhandled for now # channel 12, WCT1, unhandled for now # channel 13, WCT1, unhandled for now # channel 14, WCT1, unhandled for now # channel 15, WCT1, unhandled for now # channel 16, WCT1, unhandled for now # channel 17, WCT1, unhandled for now # channel 18, WCT1, unhandled for now # channel 19, WCT1, unhandled for now # channel 20, WCT1, unhandled for now # channel 21, WCT1, unhandled for now # channel 22, WCT1, unhandled for now # channel 23, WCT1, unhandled for now # channel 24, WCT1, unhandled for now # Global data span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #fxsks=1 loadzone = us defaultzone = us ----------------------------- -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000
O'Connor, Jonathan
2005-Dec-19 07:59 UTC
[Asterisk-Users] Can't call out on ZAP channel - need help
Michael, Does the zttool program show the PRI as working correctly? Can the PBX push calls into the Asterisk system? Also, what type of PBX is it, and is it providing the clock etc.. For the T1 connection? -Jonathan Jonathan O'Connor Senior System Administrator Inoveris LLC Direct Line (614) 791-3742 Fax (614) 791-3748 Helpdesk 866-456-1566> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Michael Sampson > Sent: Monday, December 19, 2005 9:30 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Can't call out on ZAP channel - need help > > I'm trying to connect to another PBX via an T-1 interface. I > have a T100P card. > On the CLI I get the error "Everyone is busy/congested at > this time (1:0/0/1)" When I try to dial out of the T-1 line > from an SIP softphone. > > I have posted this question a few times here and at the > asterisk forum, but can't get anyone to respond. I've seen > other people on forums with the same problem but no one has > ever given much of a solution. Does someone at least know > what the next step in debugging this problem would be. > > In the file /var/log/asterisk/full I get the error "Unable to > create channel of type 'ZAP'" > > Here are my configs. > > Zapata.conf > ------------------ > ; > ; Zapata telephony interface > ; > ; Configuration file > > [trunkgroups] > > [channels] > > language=en > context=from-pstn > ;signalling=fxs_ks > signalling=pri_net ; pri_cpe= PRI slave ; pri_net = PRI > master switchtype=qsig pridialplan=local resetinterval=never > ;rxwink=300 ; Atlas seems to use long (250ms) winks > ; > ; Whether or not to do distinctive ring detection on FXO > lines ; ;usedistinctiveringdetection=yes callerid=asreceived > usecallerid=yes hidecallerid=no callwaiting=yes > usecallingpres=yes callwaitingcallerid=yes > threewaycalling=yes transfer=yes cancallforward=yes > callreturn=yes echocancel=yes echocancelwhenbridged=yes > echotraining=400 rxgain=0.0 txgain=0.0 > group=1 > callgroup=1 > pickupgroup=1 > immediate=no > > ;faxdetect=both > faxdetect=incoming > ;faxdetect=outgoing > ;faxdetect=no > > ;Include genzaptelconf configs > #include zapata-auto.conf > > ;Include AMP configs > #include zapata_additional.conf > > channel => 1-23 > > > > ------------------------- > > > Zaptel.conf > ------------------------- > # Autogenerated by /usr/local/sbin/genzaptelconf -- do not > hand edit # Zaptel Configuration File # # This file is parsed > by the Zaptel Configurator, ztcfg # > > # It must be in the module loading order > > > # Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0" > # channel 1, WCT1, unhandled for now > # channel 2, WCT1, unhandled for now > # channel 3, WCT1, unhandled for now > # channel 4, WCT1, unhandled for now > # channel 5, WCT1, unhandled for now > # channel 6, WCT1, unhandled for now > # channel 7, WCT1, unhandled for now > # channel 8, WCT1, unhandled for now > # channel 9, WCT1, unhandled for now > # channel 10, WCT1, unhandled for now > # channel 11, WCT1, unhandled for now > # channel 12, WCT1, unhandled for now > # channel 13, WCT1, unhandled for now > # channel 14, WCT1, unhandled for now > # channel 15, WCT1, unhandled for now > # channel 16, WCT1, unhandled for now > # channel 17, WCT1, unhandled for now > # channel 18, WCT1, unhandled for now > # channel 19, WCT1, unhandled for now > # channel 20, WCT1, unhandled for now > # channel 21, WCT1, unhandled for now > # channel 22, WCT1, unhandled for now > # channel 23, WCT1, unhandled for now > # channel 24, WCT1, unhandled for now > > # Global data > > span=1,1,0,esf,b8zs > bchan=1-23 > dchan=24 > > #fxsks=1 > loadzone = us > defaultzone = us > ----------------------------- > > -- > Michael Sampson > Information Systems Manager > Customer Contact Services > msampson@yourccsteam.com > 952-936-4000 > > _______________________________________________ > --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 >
O'Connor, Jonathan
2005-Dec-19 08:29 UTC
[Asterisk-Users] Can't call out on ZAP channel - need help
The parameter in zaptel.conf that sets up timing etc is: span=1,1,0,esf,b8zs The first 1 means this is span 1. The second one defines the timing of the link. For asterisk to provide the timing use 0 instead. For instance my Asterisk box, hooked directly to my Avaya G3 uses: span=1,0,0,esf,b8zs Also, switchtype=qsig This is something I have never personally got working to any useful amount with our Definity. I use switchtype=national It doesnt have some of the features of qsig, but will get you going if the PBX is setup to use a standard National ISDN 2 switch. You will I beleive need to shut down asterisk and then run ztcfg -vvvv if you make these changes, then restart asterisk. signalling=pri_net merely makes the Asterisk box act like the telco, as far as its signaling is concerned, quite normal when hooked to a legecy pbx. Hope this helps, am no expert, just going on what I got mine running with :) -Jonathan Jonathan O'Connor Senior System Administrator Inoveris LLC Direct Line (614) 791-3742 Fax (614) 791-3748 Helpdesk 866-456-1566 ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Sampson Sent: Monday, December 19, 2005 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can't call out on ZAP channel - need help Yeah, the zttool program shows the PRI as having No Alarms. It is an Infinity system by Amtelco. I haven't actually tried making a call from the other pbx, but I did have my vendor (Amtelco) look at it and they verified that the span was up and everything was working correctly. The asterisk system is set to signalling=pri_net which I assumed meant that the asterisk box would be handling the timing. Here is the output from "pri show span 1" asterisk1*CLI> pri show span 1 Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: Q.SIG switch Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000 O'Connor, Jonathan wrote: Michael, Does the zttool program show the PRI as working correctly? Can the PBX push calls into the Asterisk system? Also, what type of PBX is it, and is it providing the clock etc.. For the T1 connection? -Jonathan Jonathan O'Connor Senior System Administrator Inoveris LLC Direct Line (614) 791-3742 Fax (614) 791-3748 Helpdesk 866-456-1566 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Sampson Sent: Monday, December 19, 2005 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Can't call out on ZAP channel - need help I'm trying to connect to another PBX via an T-1 interface. I have a T100P card. On the CLI I get the error "Everyone is busy/congested at this time (1:0/0/1)" When I try to dial out of the T-1 line from an SIP softphone. I have posted this question a few times here and at the asterisk forum, but can't get anyone to respond. I've seen other people on forums with the same problem but no one has ever given much of a solution. Does someone at least know what the next step in debugging this problem would be. In the file /var/log/asterisk/full I get the error "Unable to create channel of type 'ZAP'" Here are my configs. Zapata.conf ------------------ ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn ;signalling=fxs_ks signalling=pri_net ; pri_cpe= PRI slave ; pri_net = PRI master switchtype=qsig pridialplan=local resetinterval=never ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf channel => 1-23 ------------------------- Zaptel.conf ------------------------- # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0" # channel 1, WCT1, unhandled for now # channel 2, WCT1, unhandled for now # channel 3, WCT1, unhandled for now # channel 4, WCT1, unhandled for now # channel 5, WCT1, unhandled for now # channel 6, WCT1, unhandled for now # channel 7, WCT1, unhandled for now # channel 8, WCT1, unhandled for now # channel 9, WCT1, unhandled for now # channel 10, WCT1, unhandled for now # channel 11, WCT1, unhandled for now # channel 12, WCT1, unhandled for now # channel 13, WCT1, unhandled for now # channel 14, WCT1, unhandled for now # channel 15, WCT1, unhandled for now # channel 16, WCT1, unhandled for now # channel 17, WCT1, unhandled for now # channel 18, WCT1, unhandled for now # channel 19, WCT1, unhandled for now # channel 20, WCT1, unhandled for now # channel 21, WCT1, unhandled for now # channel 22, WCT1, unhandled for now # channel 23, WCT1, unhandled for now # channel 24, WCT1, unhandled for now # Global data span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #fxsks=1 loadzone = us defaultzone = us ----------------------------- -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000 _______________________________________________ --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... 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O'Connor, Jonathan
2005-Dec-19 09:13 UTC
[Asterisk-Users] Can't call out on ZAP channel - need help
The only other thing I can think of is that your contexts etc... need checked. It would be very helpful to know if calls can come into the system from the PBX, that would be the only way to know the span is alive and well truely. Once you know that then its down to the contexts and configs... Jonathan O'Connor Senior System Administrator Inoveris LLC Direct Line (614) 791-3742 Fax (614) 791-3748 Helpdesk 866-456-1566 ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Sampson Sent: Monday, December 19, 2005 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can't call out on ZAP channel - need help My other pbx vendor told me they supported pretty much all of the switchtypes and that the system would automatically detect the correct one. I've tried Qsig and National and both seem to bring the span up fine. I just switched to span=1,0,0,esf,b8zs to have asterisk provide the timing. That didn't change any of the errors I'm getting. So I changed the switchtype to national just to be sure, and it still didn't fix anything. Everything seems to indicate that the span is up and running fine. Any more ideas? Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000 O'Connor, Jonathan wrote: The parameter in zaptel.conf that sets up timing etc is: span=1,1,0,esf,b8zs The first 1 means this is span 1. The second one defines the timing of the link. For asterisk to provide the timing use 0 instead. For instance my Asterisk box, hooked directly to my Avaya G3 uses: span=1,0,0,esf,b8zs Also, switchtype=qsig This is something I have never personally got working to any useful amount with our Definity. I use switchtype=national It doesnt have some of the features of qsig, but will get you going if the PBX is setup to use a standard National ISDN 2 switch. You will I beleive need to shut down asterisk and then run ztcfg -vvvv if you make these changes, then restart asterisk. signalling=pri_net merely makes the Asterisk box act like the telco, as far as its signaling is concerned, quite normal when hooked to a legecy pbx. Hope this helps, am no expert, just going on what I got mine running with :) -Jonathan Jonathan O'Connor Senior System Administrator Inoveris LLC Direct Line (614) 791-3742 Fax (614) 791-3748 Helpdesk 866-456-1566 ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Sampson Sent: Monday, December 19, 2005 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can't call out on ZAP channel - need help Yeah, the zttool program shows the PRI as having No Alarms. It is an Infinity system by Amtelco. I haven't actually tried making a call from the other pbx, but I did have my vendor (Amtelco) look at it and they verified that the span was up and everything was working correctly. The asterisk system is set to signalling=pri_net which I assumed meant that the asterisk box would be handling the timing. Here is the output from "pri show span 1" asterisk1*CLI> pri show span 1 Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: Q.SIG switch Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000 O'Connor, Jonathan wrote: Michael, Does the zttool program show the PRI as working correctly? Can the PBX push calls into the Asterisk system? Also, what type of PBX is it, and is it providing the clock etc.. For the T1 connection? -Jonathan Jonathan O'Connor Senior System Administrator Inoveris LLC Direct Line (614) 791-3742 Fax (614) 791-3748 Helpdesk 866-456-1566 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Sampson Sent: Monday, December 19, 2005 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Can't call out on ZAP channel - need help I'm trying to connect to another PBX via an T-1 interface. I have a T100P card. On the CLI I get the error "Everyone is busy/congested at this time (1:0/0/1)" When I try to dial out of the T-1 line from an SIP softphone. I have posted this question a few times here and at the asterisk forum, but can't get anyone to respond. I've seen other people on forums with the same problem but no one has ever given much of a solution. Does someone at least know what the next step in debugging this problem would be. In the file /var/log/asterisk/full I get the error "Unable to create channel of type 'ZAP'" Here are my configs. Zapata.conf ------------------ ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn ;signalling=fxs_ks signalling=pri_net ; pri_cpe= PRI slave ; pri_net = PRI master switchtype=qsig pridialplan=local resetinterval=never ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf channel => 1-23 ------------------------- Zaptel.conf ------------------------- # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0" # channel 1, WCT1, unhandled for now # channel 2, WCT1, unhandled for now # channel 3, WCT1, unhandled for now # channel 4, WCT1, unhandled for now # channel 5, WCT1, unhandled for now # channel 6, WCT1, unhandled for now # channel 7, WCT1, unhandled for now # channel 8, WCT1, unhandled for now # channel 9, WCT1, unhandled for now # channel 10, WCT1, unhandled for now # channel 11, WCT1, unhandled for now # channel 12, WCT1, unhandled for now # channel 13, WCT1, unhandled for now # channel 14, WCT1, unhandled for now # channel 15, WCT1, unhandled for now # channel 16, WCT1, unhandled for now # channel 17, WCT1, unhandled for now # channel 18, WCT1, unhandled for now # channel 19, WCT1, unhandled for now # channel 20, WCT1, unhandled for now # channel 21, WCT1, unhandled for now # channel 22, WCT1, unhandled for now # channel 23, WCT1, unhandled for now # channel 24, WCT1, unhandled for now # Global data span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #fxsks=1 loadzone = us defaultzone = us ----------------------------- -- Michael Sampson Information Systems Manager Customer Contact Services msampson@yourccsteam.com 952-936-4000 _______________________________________________ --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 ________________________________ _______________________________________________ --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... 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