Re: [Asterisk-Users] Channel 0/1, span 1 got hangup request
Steve Totaro wrote: What are you doing in between making changes and testing the changes? After changing settings I reboot system! Really. :) Because other actions have no effect. Also reboot, too.. Thanks, Steve Just a couple guesses on things to try. Zapata.conf 1. Changing switchtype variables (doubtful but give it a try). 2. Add a variable to define pridialplan (I remember someone setting this to unknown to solve a similar issue) Try pridialplan=unknown and/or prilocaldialplan=local or some other valid option. A do this config, but no effects Zaptel.conf 1. span=1,1,5,ccs,hdb3 I think that your dial statement or the pridialplan is your issue. If you look at the debug info Here is something suspicious: -- Called g1/100 unless 100 is the number you are trying to dial outbound. If the above fails, then try below. Try tweaking your settings here like span=1,0,0,ccs,hdb3 What is the provider expecting? No effect on settings: span=1,0,0,ccs,hdb3 span=1,1,5,ccs,hdb3 span=1,2,4,ccs,hdb3 Thanks, Steve Dear Users, I have an Digium Wildcard TE110P T1/E1 Card inserted in Linux box runnig Asterisk 1.2.0 All incoming calls from E1 interface to SIP-phone goes exellent, but calls from SIP to E1 gives the errors: -- Executing Dial(SIP/anton-6cf4, Zap/g1/100) in new stack -- Making new call for cr 32775 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 05 41 6e 74 6f 6e] Display (len= 5) +)[EMAIL PROTECTED]@[EMAIL PROTECTED]@│@[ Anton ] [6c 0d 21 81 38 34 37 37 33 36 31 38 31 38 33] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '84773618183' ] [70 04 a1 31 30 30] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ] -- Called g1/100 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup request Dec 5 15:30:12 WARNING[30946]: app_dial.c:706 wait_for_answer: Unable to forward voice NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == No one is available to answer at this time (1:0/0/0) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 80 d1] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (81), class = Invalid message (5) ] -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Timeout on SIP/anton-6cf4 == CDR updated on SIP/anton-6cf4 -- Executing Hangup(SIP/anton-6cf4, ) in new stack /etc/zaptel.conf span=1,1,5,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = nl defaultzone=nl /etc/asterisck/zapata.conf [trunkgroups] [channels] language=en signalling=pri_cpe switchtype=euroisdn echocancel=32 echocancelwhenbridged=yes usecallerid=yes callerid=asreceived transfer=yes overlapdial=yes cancallforward=yes group=1 context=zapata channel = 1-15,17-31 Has anybody resolve this problem? -- SY, Anton V Bakulev. MIPT-telecom. [EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] Channel 0/1, span 1 got hangup request
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, December 10, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Channel 0/1, span 1 got hangup request Just a couple guesses on things to try. Zapata.conf 1. Changing switchtype variables (doubtful but give it a try). 2. Add a variable to define pridialplan (I remember someone setting this to unknown to solve a similar issue) Try pridialplan=unknown and/or prilocaldialplan=local or some other valid option. A do this config, but no effects Zaptel.conf 1. span=1,1,5,ccs,hdb3 I think that your dial statement or the pridialplan is your issue. If you look at the debug info Here is something suspicious: -- Called g1/100 unless 100 is the number you are trying to dial outbound. If the above fails, then try below. Try tweaking your settings here like span=1,0,0,ccs,hdb3 What is the provider expecting? No effect on settings: span=1,0,0,ccs,hdb3 span=1,1,5,ccs,hdb3 span=1,2,4,ccs,hdb3 Thanks, Steve Dear Users, I have an Digium Wildcard TE110P T1/E1 Card inserted in Linux box runnig Asterisk 1.2.0 All incoming calls from E1 interface to SIP-phone goes exellent, but calls from SIP to E1 gives the errors: -- Executing Dial(SIP/anton-6cf4, Zap/g1/100) in new stack -- Making new call for cr 32775 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 05 41 6e 74 6f 6e] Display (len= 5) +)[EMAIL PROTECTED]@[EMAIL PROTECTED]@│@[ Anton ] [6c 0d 21 81 38 34 37 37 33 36 31 38 31 38 33] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '84773618183' ] [70 04 a1 31 30 30] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ] -- Called g1/100 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup request Dec 5 15:30:12 WARNING[30946]: app_dial.c:706 wait_for_answer: Unable to forward voice NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == No one is available to answer at this time (1:0/0/0) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 80 d1] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (81), class = Invalid message (5) ] -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Timeout on SIP/anton-6cf4 == CDR updated on SIP/anton-6cf4 -- Executing Hangup(SIP/anton-6cf4, ) in new stack /etc/zaptel.conf span=1,1,5,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = nl defaultzone=nl /etc/asterisck/zapata.conf [trunkgroups] [channels] language=en signalling=pri_cpe switchtype=euroisdn echocancel=32 echocancelwhenbridged=yes usecallerid=yes callerid=asreceived transfer=yes overlapdial=yes cancallforward=yes group=1 context=zapata channel = 1-15,17-31 Has anybody resolve this problem? -- SY, Anton V Bakulev. MIPT-telecom. [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk
RE: [Asterisk-Users] Channel 0/1, span 1 got hangup request
What are you doing in between making changes and testing the changes? Thanks, Steve Just a couple guesses on things to try. Zapata.conf 1. Changing switchtype variables (doubtful but give it a try). 2. Add a variable to define pridialplan (I remember someone setting this to unknown to solve a similar issue) Try pridialplan=unknown and/or prilocaldialplan=local or some other valid option. A do this config, but no effects Zaptel.conf 1. span=1,1,5,ccs,hdb3 I think that your dial statement or the pridialplan is your issue. If you look at the debug info Here is something suspicious: -- Called g1/100 unless 100 is the number you are trying to dial outbound. If the above fails, then try below. Try tweaking your settings here like span=1,0,0,ccs,hdb3 What is the provider expecting? No effect on settings: span=1,0,0,ccs,hdb3 span=1,1,5,ccs,hdb3 span=1,2,4,ccs,hdb3 Thanks, Steve Dear Users, I have an Digium Wildcard TE110P T1/E1 Card inserted in Linux box runnig Asterisk 1.2.0 All incoming calls from E1 interface to SIP-phone goes exellent, but calls from SIP to E1 gives the errors: -- Executing Dial(SIP/anton-6cf4, Zap/g1/100) in new stack -- Making new call for cr 32775 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 05 41 6e 74 6f 6e] Display (len= 5) +)[EMAIL PROTECTED]@[EMAIL PROTECTED]@│@[ Anton ] [6c 0d 21 81 38 34 37 37 33 36 31 38 31 38 33] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '84773618183' ] [70 04 a1 31 30 30] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ] -- Called g1/100 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup request Dec 5 15:30:12 WARNING[30946]: app_dial.c:706 wait_for_answer: Unable to forward voice NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == No one is available to answer at this time (1:0/0/0) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 80 d1] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (81), class = Invalid message (5) ] -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Timeout on SIP/anton-6cf4 == CDR updated on SIP/anton-6cf4 -- Executing Hangup(SIP/anton-6cf4, ) in new stack /etc/zaptel.conf span=1,1,5,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = nl defaultzone=nl /etc/asterisck/zapata.conf [trunkgroups] [channels] language=en signalling=pri_cpe switchtype=euroisdn echocancel=32 echocancelwhenbridged=yes usecallerid=yes callerid=asreceived transfer=yes overlapdial=yes cancallforward=yes group=1 context=zapata channel = 1-15,17-31 Has anybody resolve this problem? -- SY, Anton V Bakulev. MIPT-telecom. [EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] Channel 0/1, span 1 got hangup request
Just a couple guesses on things to try. Zapata.conf 1. Changing switchtype variables (doubtful but give it a try). 2. Add a variable to define pridialplan (I remember someone setting this to unknown to solve a similar issue) Try pridialplan=unknown and/or prilocaldialplan=local or some other valid option. Zaptel.conf 1. span=1,1,5,ccs,hdb3 I think that your dial statement or the pridialplan is your issue. If you look at the debug info Here is something suspicious: -- Called g1/100 unless 100 is the number you are trying to dial outbound. If the above fails, then try below. Try tweaking your settings here like span=1,0,0,ccs,hdb3 What is the provider expecting? Thanks, Steve Dear Users, I have an Digium Wildcard TE110P T1/E1 Card inserted in Linux box runnig Asterisk 1.2.0 All incoming calls from E1 interface to SIP-phone goes exellent, but calls from SIP to E1 gives the errors: -- Executing Dial(SIP/anton-6cf4, Zap/g1/100) in new stack -- Making new call for cr 32775 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 05 41 6e 74 6f 6e] Display (len= 5) ╫)[EMAIL PROTECTED]@[EMAIL PROTECTED]@│@[ Anton ] [6c 0d 21 81 38 34 37 37 33 36 31 38 31 38 33] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '84773618183' ] [70 04 a1 31 30 30] Called Number (len= 6) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ] -- Called g1/100 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup request Dec 5 15:30:12 WARNING[30946]: app_dial.c:706 wait_for_answer: Unable to forward voice NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == No one is available to answer at this time (1:0/0/0) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 80 d1] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (81), class = Invalid message (5) ] -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Timeout on SIP/anton-6cf4 == CDR updated on SIP/anton-6cf4 -- Executing Hangup(SIP/anton-6cf4, ) in new stack /etc/zaptel.conf span=1,1,5,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = nl defaultzone=nl /etc/asterisck/zapata.conf [trunkgroups] [channels] language=en signalling=pri_cpe switchtype=euroisdn echocancel=32 echocancelwhenbridged=yes usecallerid=yes callerid=asreceived transfer=yes overlapdial=yes cancallforward=yes group=1 context=zapata channel = 1-15,17-31 Has anybody resolve this problem? -- SY, Anton V Bakulev. MIPT-telecom. [EMAIL PROTECTED] ___ --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
