Jens Hansen
2004-Nov-11 14:12 UTC
[Asterisk-Users] sometimes problem with dialing ZAP channel
On about one call of 10 calls i am getting this error when there is a call coming in and the ZAP device is not ringing: --------------------------------------------------> [6c 02 00 c3] > Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:Unknown Number Plan (0)> Presentation: Number not available (67) '' ] > [70 09 c1 34 30 31 36 37 37 39 31] > Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI:ISDN/Telephony Numbering Plan (E.164/E.163) (1) '40167791' ]> [a1] > Sending Complete (len= 1)-- Making new call for cr 130> Protocol Discriminator: Q.931 (8) len=28 > Call Ref: len= 1 (reference 2/0x2) (Originator) > Message type: SETUP (5) > [04 03 80 90 a3] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfercapability: Speech (0)> Ext: 1 Trans mode/rate: 64kbps, circuit-mode(16)> Ext: 1 User information layer 1: A-Law (35) > [18 01 89] > Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, ExclusiveDchan: 0> ChanSel: B1 channel]> [6c 02 00 c3] > Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:Unknown Number Plan (0)> Presentation: Number not available (67) '' ] > [70 09 c1 34 30 31 36 37 37 39 31] > Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI:ISDN/Telephony Numbering Plan (E.164/E.163) (1) '40167791' ]> [a1] > Sending Complete (len= 1)-- Called g1/40167791 Nov 11 21:49:14 NOTICE[344085]: app_dial.c:756 dial_exec: Unable to create channel of type 'SIP' No response to SETUP message No response to SETUP message NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending No response to SETUP message NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending -- Channel 0/1, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending -- Hungup 'Zap/1-1' == No one is available to answer at this time ------------------- i am using the following setup: ISDN (PTP) <-> AVM FRITZ CARD <-> ASTERISK <-> HFC CARD <-> ISDN phone here is my zapata.conf: --------------------- [channels] switchtype = euroisdn signalling = bri_net_ptmp pridialplan = local prilocaldialplan = local echocancel = yes overlapdial = no immediate = no usecallerid = yes group = 1 context = isdn2out usecallingpres=yes musiconhold=default channel => 1-2 threewaycalling=yes transfer=yes ----------------------- Someone please give me a hint i am already 2 days at this problem and i am not able to solve it. Thanks Jens
Jens Hansen
2004-Nov-11 14:20 UTC
[Asterisk-Users] sometimes problem with dialing ZAP channel
forgot half of it: -------------------------------- extensions.conf [jens-privat] exten => s,1,Wait(0.2) exten => s,2,Dial(Zap/g1/40167791&SIP/20,40,t) exten => s,3,Voicemail(u40167791) exten => s,4,Hangup exten => s,103,Hangup -------------------------------- Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 (a) Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jens Hansen Sent: Thursday, November 11, 2004 10:13 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] sometimes problem with dialing ZAP channel On about one call of 10 calls i am getting this error when there is a call coming in and the ZAP device is not ringing: --------------------------------------------------> [6c 02 00 c3] > Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:Unknown Number Plan (0)> Presentation: Number not available (67) '' ] > [70 09 c1 34 30 31 36 37 37 39 31] > Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI:ISDN/Telephony Numbering Plan (E.164/E.163) (1) '40167791' ]> [a1] > Sending Complete (len= 1)-- Making new call for cr 130> Protocol Discriminator: Q.931 (8) len=28 > Call Ref: len= 1 (reference 2/0x2) (Originator) > Message type: SETUP (5) > [04 03 80 90 a3] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfercapability: Speech (0)> Ext: 1 Trans mode/rate: 64kbps, circuit-mode(16)> Ext: 1 User information layer 1: A-Law (35) > [18 01 89] > Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, ExclusiveDchan: 0> ChanSel: B1 channel]> [6c 02 00 c3] > Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:Unknown Number Plan (0)> Presentation: Number not available (67) '' ] > [70 09 c1 34 30 31 36 37 37 39 31] > Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI:ISDN/Telephony Numbering Plan (E.164/E.163) (1) '40167791' ]> [a1] > Sending Complete (len= 1)-- Called g1/40167791 Nov 11 21:49:14 NOTICE[344085]: app_dial.c:756 dial_exec: Unable to create channel of type 'SIP' No response to SETUP message No response to SETUP message NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending No response to SETUP message NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending -- Channel 0/1, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending -- Hungup 'Zap/1-1' == No one is available to answer at this time ------------------- i am using the following setup: ISDN (PTP) <-> AVM FRITZ CARD <-> ASTERISK <-> HFC CARD <-> ISDN phone here is my zapata.conf: --------------------- [channels] switchtype = euroisdn signalling = bri_net_ptmp pridialplan = local prilocaldialplan = local echocancel = yes overlapdial = no immediate = no usecallerid = yes group = 1 context = isdn2out usecallingpres=yes musiconhold=default channel => 1-2 threewaycalling=yes transfer=yes ----------------------- Someone please give me a hint i am already 2 days at this problem and i am not able to solve it. Thanks Jens _______________________________________________ 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