Mike Flynn
2005-Mar-28 13:04 UTC
[Asterisk-Users] 8 channel fxo setup outgoing call problem
I have an eight channel fxo setup (2 TDM400P cards) and I have them setup. Here are my configs: Zaptel.conf: fxsks=1-8 loadzone=us defaultzone=us Zapata.conf: [trunkgroups] [channels] musiconhold=default rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callerid=xxxxxx callwaiting=yes busydetect=no callprogress=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=incoming faxdetect=no signalling=fxs_ks callerid=asreceived context=from-pstn channel=>1-8 Now calling in is no problem, all the channels pick up the incoming call just fine. However, for some reason, calling out does not work. When I dial out (I have a default trunk setup) I get this on the console: -- Executing Macro("SIP/201-1d90", "dialout-default|636399xxx") in new stack -- Executing GotoIf("SIP/201-1d90", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/201-1d90", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing Dial("SIP/201-1d90", "ZAP/g0/6363997681") in new stack -- Called g0/636399xxxx -- Zap/1-1 answered SIP/201-1d90 But it doesn't answer, nothing rings (locally or on my cellphone, the test number I'm calling out to) it just says "connected" on the local phone but its not actually connecting. This problem is very weird because as of 5 hours ago, I only had 1 TDM400P card and 1 FXO chip and I could make outgoing calls with no problems at all. P.S. I have checked the phone line I'm testing this on and it is fine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050328/77d851bf/attachment.htm
Goutam Shaw
2005-Mar-28 13:15 UTC
[Asterisk-Users] 8 channel fxo setup outgoing call problem
Do you have ATI FXO daughter cards. We had experienced similar problems. After replacing the ATI with Digium X100M Rev B daughter cards the system has been running fine. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Mike Flynn Sent: March 28, 2005 3:05 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 8 channel fxo setup outgoing call problem I have an eight channel fxo setup (2 TDM400P cards) and I have them setup. Here are my configs: Zaptel.conf: fxsks=1-8 loadzone=us defaultzone=us Zapata.conf: [trunkgroups] [channels] musiconhold=default rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callerid=xxxxxx callwaiting=yes busydetect=no callprogress=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=incoming faxdetect=no signalling=fxs_ks callerid=asreceived context=from-pstn channel=>1-8 Now calling in is no problem, all the channels pick up the incoming call just fine. However, for some reason, calling out does not work. When I dial out (I have a default trunk setup) I get this on the console: -- Executing Macro("SIP/201-1d90", "dialout-default|636399xxx") in new stack -- Executing GotoIf("SIP/201-1d90", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/201-1d90", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing Dial("SIP/201-1d90", "ZAP/g0/6363997681") in new stack -- Called g0/636399xxxx -- Zap/1-1 answered SIP/201-1d90 But it doesn't answer, nothing rings (locally or on my cellphone, the test number I'm calling out to) it just says "connected" on the local phone but its not actually connecting. This problem is very weird because as of 5 hours ago, I only had 1 TDM400P card and 1 FXO chip and I could make outgoing calls with no problems at all. P.S. I have checked the phone line I'm testing this on and it is fine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050328/bb7b60d9/attachment.htm