Re: [asterisk-users] Outgoing call failure
Hi all, My issue was resolved. It was an issue with service provider. I got help from some smart guys, they have helped me a lot to setup my PRI up and running. Thank you very much. Sample link may helpful to identify the simmilar issues : http://networking.ringofsaturn.com/Routers/isdncausecodes.php I have removed crc4 from my /etc/dahdi/system.conf file, because BSNL in India not using crc4. Michael.k On Wed, Oct 19, 2011 at 10:43 AM, michael k michael.in...@gmail.com wrote: Hi List, My all incoming calls are working fine but i cant make outgoing calls. There was no issues for both incoming and outgoing calls till yesterday. Can somebody tell me what is the issue is ?. I have enable the dibugging in pri line by issue the command pri set debug on span 1 on CLI. *The output of my outgoing call * -- Executing [s@macro-outbound-callerid:12] ExecIf(SIP/2015-0012, 1?Set(CALLERID(all)=08023515000)) in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf(SIP/2015-0012, 0?Set(CALLERID(all)=)) in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf(SIP/2015-0012, 0?Set(CALLERID(all)=)) in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf(SIP/2015-0012, 0?Set(CALLERPRES()=prohib_passed_screen)) in new stack -- Executing [s@macro-dialout-trunk:12] ExecIf(SIP/2015-0012, 0?AGI(fixlocalprefix)) in new stack -- Executing [s@macro-dialout-trunk:13] Set(SIP/2015-0012, OUTNUM=9741735245) in new stack -- Executing [s@macro-dialout-trunk:14] Set(SIP/2015-0012, custom=DAHDI/g0) in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf(SIP/2015-0012, 0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))) in new stack -- Executing [s@macro-dialout-trunk:16] Macro(SIP/2015-0012, dialout-trunk-predial-hook,) in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/2015-0012, ) in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf(SIP/2015-0012, 0?bypass,1) in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf(SIP/2015-0012, 0?customtrunk) in new stack -- Executing [s@macro-dialout-trunk:19] Dial(SIP/2015-0012, DAHDI/g0/9741735245,300,) in new stack -- Making new call for cref 32780 -- Requested transfer capability: 0x00 - SPEECH DL-DATA request Protocol Discriminator: Q.931 (8) len=44 TEI=0 Call Ref: len= 2 (reference 12/0xC) (Sent from originator) Message Type: SETUP (5) TEI=0 Transmitting N(S)=28, window is open V(A)=28 K=7 Protocol Discriminator: Q.931 (8) len=44 TEI=0 Call Ref: len= 2 (reference 12/0xC) (Sent from 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) User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: CPE] [6c 0d 21 81 30 38 30 32 33 35 31 35 30 30 30] 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) '08023515000' ] [70 0b a1 39 37 34 31 37 33 35 32 34 35] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9741735245' ] [a1] Sending Complete (len= 1) q931.c:5039 q931_setup: Call 32780 enters state 1 (Call Initiated). Hold state: Idle -- Called g0/9741735245 Protocol Discriminator: Q.931 (8) len=25 TEI=0 Call Ref: len= 2 (reference 12/0xC) (Sent to originator) Message Type: RELEASE COMPLETE (90) [08 03 02 80 95] Cause (len= 5) [ Ext: 0 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unknown (0), class = Normal Event (0) ] Cause data 1: 95 (149) [28 0d 43 41 4c 4c 20 52 45 4a 45 43 54 45 44] Display (len=13) [ CALL REJECTED ] Received message for call 0x2c108060 on 0x2c0dc220 TEI/SAPI 0/0, call-pri is 0x2c0dc220 TEI/SAPI 0/0 -- Processing IE 8 (cs0, Cause) -- Processing IE 40 (cs0, Display) q931.c:7197 post_handle_q931_message: Call 32780 enters state 0 (Null). Hold state: Idle -- Span 1: Channel 0/1 got hangup, cause 0 q931_hangup: other hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state Idle NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-state Idle -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0
[asterisk-users] Outgoing call failure
Hi List, My all incoming calls are working fine but i cant make outgoing calls. There was no issues for both incoming and outgoing calls till yesterday. Can somebody tell me what is the issue is ?. I have enable the dibugging in pri line by issue the command pri set debug on span 1 on CLI. *The output of my outgoing call * -- Executing [s@macro-outbound-callerid:12] ExecIf(SIP/2015-0012, 1?Set(CALLERID(all)=08023515000)) in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf(SIP/2015-0012, 0?Set(CALLERID(all)=)) in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf(SIP/2015-0012, 0?Set(CALLERID(all)=)) in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf(SIP/2015-0012, 0?Set(CALLERPRES()=prohib_passed_screen)) in new stack -- Executing [s@macro-dialout-trunk:12] ExecIf(SIP/2015-0012, 0?AGI(fixlocalprefix)) in new stack -- Executing [s@macro-dialout-trunk:13] Set(SIP/2015-0012, OUTNUM=9741735245) in new stack -- Executing [s@macro-dialout-trunk:14] Set(SIP/2015-0012, custom=DAHDI/g0) in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf(SIP/2015-0012, 0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))) in new stack -- Executing [s@macro-dialout-trunk:16] Macro(SIP/2015-0012, dialout-trunk-predial-hook,) in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/2015-0012, ) in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf(SIP/2015-0012, 0?bypass,1) in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf(SIP/2015-0012, 0?customtrunk) in new stack -- Executing [s@macro-dialout-trunk:19] Dial(SIP/2015-0012, DAHDI/g0/9741735245,300,) in new stack -- Making new call for cref 32780 -- Requested transfer capability: 0x00 - SPEECH DL-DATA request Protocol Discriminator: Q.931 (8) len=44 TEI=0 Call Ref: len= 2 (reference 12/0xC) (Sent from originator) Message Type: SETUP (5) TEI=0 Transmitting N(S)=28, window is open V(A)=28 K=7 Protocol Discriminator: Q.931 (8) len=44 TEI=0 Call Ref: len= 2 (reference 12/0xC) (Sent from 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) User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: CPE] [6c 0d 21 81 30 38 30 32 33 35 31 35 30 30 30] 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) '08023515000' ] [70 0b a1 39 37 34 31 37 33 35 32 34 35] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9741735245' ] [a1] Sending Complete (len= 1) q931.c:5039 q931_setup: Call 32780 enters state 1 (Call Initiated). Hold state: Idle -- Called g0/9741735245 Protocol Discriminator: Q.931 (8) len=25 TEI=0 Call Ref: len= 2 (reference 12/0xC) (Sent to originator) Message Type: RELEASE COMPLETE (90) [08 03 02 80 95] Cause (len= 5) [ Ext: 0 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unknown (0), class = Normal Event (0) ] Cause data 1: 95 (149) [28 0d 43 41 4c 4c 20 52 45 4a 45 43 54 45 44] Display (len=13) [ CALL REJECTED ] Received message for call 0x2c108060 on 0x2c0dc220 TEI/SAPI 0/0, call-pri is 0x2c0dc220 TEI/SAPI 0/0 -- Processing IE 8 (cs0, Cause) -- Processing IE 40 (cs0, Display) q931.c:7197 post_handle_q931_message: Call 32780 enters state 0 (Null). Hold state: Idle -- Span 1: Channel 0/1 got hangup, cause 0 q931_hangup: other hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state Idle NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-state Idle -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/2015-0012, Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0) in new stack -- Executing [s@macro-dialout-trunk:21] Goto(SIP/2015-0012, s-CHANUNAVAIL,1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(SIP/2015-0012, RC=0) in new stack -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(SIP/2015-0012, 0,1) in new stack -- Goto (macro-dialout-trunk,0,1) -- Executing [99741735245@from-internal:5]
[asterisk-users] Outgoing call failure
-- Forwarded message -- From: michael k michael.in...@gmail.com Date: Wed, Oct 19, 2011 at 10:43 AM Subject: Outgoing call failure To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi List, My all incoming calls are working fine but i cant make outgoing calls. There was no issues for both incoming and outgoing calls till yesterday. Can somebody tell me what is the issue is ?. I have enable the dibugging in pri line by issue the command pri set debug on span 1 on CLI. *The output of my outgoing call * -- Executing [s@macro-outbound-callerid:12] ExecIf(SIP/2015-0012, 1?Set(CALLERID(all)=08023515000)) in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf(SIP/2015-0012, 0?Set(CALLERID(all)=)) in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf(SIP/2015-0012, 0?Set(CALLERID(all)=)) in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf(SIP/2015-0012, 0?Set(CALLERPRES()=prohib_passed_screen)) in new stack -- Executing [s@macro-dialout-trunk:12] ExecIf(SIP/2015-0012, 0?AGI(fixlocalprefix)) in new stack -- Executing [s@macro-dialout-trunk:13] Set(SIP/2015-0012, OUTNUM=9741735245) in new stack -- Executing [s@macro-dialout-trunk:14] Set(SIP/2015-0012, custom=DAHDI/g0) in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf(SIP/2015-0012, 0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))) in new stack -- Executing [s@macro-dialout-trunk:16] Macro(SIP/2015-0012, dialout-trunk-predial-hook,) in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/2015-0012, ) in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf(SIP/2015-0012, 0?bypass,1) in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf(SIP/2015-0012, 0?customtrunk) in new stack -- Executing [s@macro-dialout-trunk:19] Dial(SIP/2015-0012, DAHDI/g0/9741735245,300,) in new stack -- Making new call for cref 32780 -- Requested transfer capability: 0x00 - SPEECH DL-DATA request Protocol Discriminator: Q.931 (8) len=44 TEI=0 Call Ref: len= 2 (reference 12/0xC) (Sent from originator) Message Type: SETUP (5) TEI=0 Transmitting N(S)=28, window is open V(A)=28 K=7 Protocol Discriminator: Q.931 (8) len=44 TEI=0 Call Ref: len= 2 (reference 12/0xC) (Sent from 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) User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: CPE] [6c 0d 21 81 30 38 30 32 33 35 31 35 30 30 30] 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) '08023515000' ] [70 0b a1 39 37 34 31 37 33 35 32 34 35] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9741735245' ] [a1] Sending Complete (len= 1) q931.c:5039 q931_setup: Call 32780 enters state 1 (Call Initiated). Hold state: Idle -- Called g0/9741735245 Protocol Discriminator: Q.931 (8) len=25 TEI=0 Call Ref: len= 2 (reference 12/0xC) (Sent to originator) Message Type: RELEASE COMPLETE (90) [08 03 02 80 95] Cause (len= 5) [ Ext: 0 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unknown (0), class = Normal Event (0) ] Cause data 1: 95 (149) [28 0d 43 41 4c 4c 20 52 45 4a 45 43 54 45 44] Display (len=13) [ CALL REJECTED ] Received message for call 0x2c108060 on 0x2c0dc220 TEI/SAPI 0/0, call-pri is 0x2c0dc220 TEI/SAPI 0/0 -- Processing IE 8 (cs0, Cause) -- Processing IE 40 (cs0, Display) q931.c:7197 post_handle_q931_message: Call 32780 enters state 0 (Null). Hold state: Idle -- Span 1: Channel 0/1 got hangup, cause 0 q931_hangup: other hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state Idle NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-state Idle -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/2015-0012, Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0) in new stack -- Executing [s@macro-dialout-trunk:21] Goto(SIP/2015-0012, s-CHANUNAVAIL,1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro
[asterisk-users] SIP Device and ZAP device
Hi List, What is the diffidence between A Generic SIP Device and Generic ZAP Device while we create an extension in FreePBX ? Mic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI E1 call termination issue
Hi List, I have configured TE121PF card in E1 mode. I am using asterisk 1.6 and freepbx 2.7. My card staus turn in to green and looks like sync with the service provider. My service provider is BSNL - India. I have one toll free number for incoming and one land line number for out going calls. Problem : If i am calling to the toll free number, i am getting the ring but that call is not reaching to my asterisk box. Both incoming and outgoing are failure. Please reffer the following informations for understand the issue further. 1. [root@localhost src]# cat /etc/dahdi/system.conf span=1,0,0,CCS,HDB3,CRC4 bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us 2. cat /etc/asterisk/chan_dahdi.conf ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf 3. cat /etc/asterisk/chan_dahdi_groups.conf ; [span_1] signalling=pri_cpe switchtype=euroisdn pridialplan=national prilocaldialplan=national group=1 context=from-pstn channel=1-15,17-31 ** *Some CLI outputs* localhost*CLI *dahdi show channel 1 *Channel: 1 File Descriptor: 14 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 1 Signalling Type: ISDN PRI Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 1 taps (unless TDM bridged) currently OFF Wait for dialtone: 0ms PRI Flags: PRI Logical Span: Implicit Hookstate (FXS only): Onhook localhost*CLI 2. localhost*CLI *pri show spans* PRI span 1/0: Provisioned, Down, Active localhost*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No 3. localhost*CLI *pri set debug on span 1* Enabled debugging on span 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) Changing from state 5(Awaiting establishment) to 4(TEI assigned) TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) TEI=0 Sending SABME Changing from state 4(TEI assigned) to 5(Awaiting establishment) TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) Changing from state 5(Awaiting establishment) to 4(TEI assigned) TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) TEI=0 Sending SABME Changing from state 4(TEI assigned) to 5(Awaiting establishment) TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) Changing from state 5(Awaiting establishment) to 4(TEI assigned) TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) TEI=0 Sending SABME Changing from state 4(TEI assigned) to 5(Awaiting establishment) TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 Sending SABME TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) Changing from state 5(Awaiting establishment) to 4(TEI assigned) TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) Please some one help me to identify the issue Michael.k -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi show status command not avilable in CLI
*[root@astrisks ~]# cat /etc/asterisk/dahdi-channels.conf* ; Autogenerated by /usr/sbin/dahdi_genconf on Thu Oct 6 18:28:14 2011 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) ;;; line=1 WCFXO/0/0 FXSLS (In use) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 1 callerid= group= context=default *[root@astrisks ~]# cat /etc/asterisk/chan_dahdi_additional.conf* ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; * [root@astrisks ~]# cat /etc/asterisk/chan_dahdi_groups.conf* ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; signalling=fxs_ls context=from-analog On Fri, Oct 7, 2011 at 11:11 PM, Sammy Govind govoi...@gmail.com wrote: Please paste the configurations in the #included files as well. On Fri, Oct 7, 2011 at 7:07 PM, michael k mich...@inapp.com wrote: Hi, This is my /etc/asterisk/chan_dahdi.conf file. [root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ;added by mic 06-oct-20011 #include /etc/asterisk/dahdi-channels.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf Any issues in this ? Michael.k On Fri, Oct 7, 2011 at 7:19 PM, Eric Wieling ewiel...@nyigc.com wrote: It is likely you have an error in your /etc/asterisk/chan_dahdi.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Friday, October 07, 2011 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I am getting this error message while executing the module load chan_dahdi.so. astrisks*CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_general.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_groups.conf': == Found == Parsing '/etc/asterisk/dahdi-channels.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found Thanks, Michael.k On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote: What happens when you do the module load chan_dahdi.so command? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, astrisks*CLI module unload chan_dahdi.so Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues
Re: [asterisk-users] dahdi show status command not avilable in CLI
Hi, I am getting this error message while executing the module load chan_dahdi.so. astrisks*CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_general.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_groups.conf': == Found == Parsing '/etc/asterisk/dahdi-channels.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found Thanks, Michael.k On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote: What happens when you do the module load chan_dahdi.so command? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, astrisks*CLI module unload chan_dahdi.so Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues ? Michael.k On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, what if you reload the module chan_dahdi from within the * CLI? It should give some hints. Giorgio On 10/06/2011 05:22 PM, michael k wrote: Hi Giorgio, Thanks for your reply. I will produce some output for your reference. # lsmod | grep dahdi dahdi_echocan_mg2 39688 1 dahdi_transcode42372 1 wctc4xxp dahdi_voicebus 79424 2 wctdm24xxp,wcte12xp dahdi 238384 14 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 35265 2 wctdm24xxp,dahdi # service dahdi status ### Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 FXOFXSKS (SWEC: MG2) (battery) # dahdi_cfg -vv DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to mg2 # dahdi_scan [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO These outputs shows that the modules are loaded correctly. Any other clues ? Michael.k On Thu, Oct 6, 2011 at 8:43 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, if you type dah followed by TAB and nothing appears, it means you do not have dahdi module loaded or dahdi_cfg application not launched before starting asterisk. Giorgio On 10/06/2011 04:57 PM, michael k wrote: Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my
Re: [asterisk-users] dahdi show status command not avilable in CLI
Hi, This is my /etc/asterisk/chan_dahdi.conf file. [root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ;added by mic 06-oct-20011 #include /etc/asterisk/dahdi-channels.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf Any issues in this ? Michael.k On Fri, Oct 7, 2011 at 7:19 PM, Eric Wieling ewiel...@nyigc.com wrote: It is likely you have an error in your /etc/asterisk/chan_dahdi.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Friday, October 07, 2011 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I am getting this error message while executing the module load chan_dahdi.so. astrisks*CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_general.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_groups.conf': == Found == Parsing '/etc/asterisk/dahdi-channels.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found Thanks, Michael.k On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote: What happens when you do the module load chan_dahdi.so command? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, astrisks*CLI module unload chan_dahdi.so Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues ? Michael.k On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, what if you reload the module chan_dahdi from within the * CLI? It should give some hints. Giorgio On 10/06/2011 05:22 PM, michael k wrote: Hi Giorgio, Thanks for your reply. I will produce some output for your reference. # lsmod | grep dahdi dahdi_echocan_mg2 39688 1 dahdi_transcode42372 1 wctc4xxp dahdi_voicebus 79424 2 wctdm24xxp,wcte12xp dahdi 238384 14 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 35265 2 wctdm24xxp,dahdi # service dahdi status ### Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 FXOFXSKS (SWEC: MG2) (battery) # dahdi_cfg -vv DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to mg2
Re: [asterisk-users] dahdi show status command not avilable in CLI
Hi, astrisks*CLI* module unload chan_dahdi.so* Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues ? Michael.k On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, what if you reload the module chan_dahdi from within the * CLI? It should give some hints. Giorgio On 10/06/2011 05:22 PM, michael k wrote: Hi Giorgio, Thanks for your reply. I will produce some output for your reference. # lsmod | grep dahdi dahdi_echocan_mg2 39688 1 dahdi_transcode42372 1 wctc4xxp dahdi_voicebus 79424 2 wctdm24xxp,wcte12xp dahdi 238384 14 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 35265 2 wctdm24xxp,dahdi # service dahdi status ### Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 FXOFXSKS (SWEC: MG2) (battery) # dahdi_cfg -vv DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to mg2 # dahdi_scan [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO These outputs shows that the modules are loaded correctly. Any other clues ? Michael.k On Thu, Oct 6, 2011 at 8:43 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, if you type dah followed by TAB and nothing appears, it means you do not have dahdi module loaded or dahdi_cfg application not launched before starting asterisk. Giorgio On 10/06/2011 04:57 PM, michael k wrote: Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my asterisk box. But in my cli mode i am not getting the command dahdi show status Output of CLI : astrisks*CLI dahdi show status No such command 'dahdi show status' (type 'core show help dahdi show' for other possible commands) I really not understand why this command is not avilable in CLI mode. Please help me to resolve this issue Michael.k -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] PSTN connectivity
Hi, Please see the sample. A ) Analog HardwareType Ports Action FXO Ports 1 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo FXS Ports -- B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog* * C ) ZAP Trunk (DAHDI compatibility Mode)* Trunk Description: Outbound Caller ID:CID Options: Maximum Channels: Disable Trunk: Disable Monitor Trunk Failures: Enable Outgoing Dial Rules Dial Rules: 0471+NXX Dial Rules Wizards: Outbound Dial Prefix:Outgoing Settings Zap Identifier (trunk name): *D ) INBOUND route * Description: Extensions: 199 * E ) **OUTBOUND Route* Route Name: 9_outside Route CID: Override Extension CID Route Password: PIN Set: Emergency Dialing: Intra Company Route: Music On Hold? Dial Patterns 8|NXXNXX 8|NXX Dial patterns wizards*: * Trunk SequenceZAP/g0 0 * F ) In command Line I can see the following things * [root@astrisks ~]# *dahdi_cfg -vv* DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to none [root@astrisks ~]# *dahdi_scan* [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO *Asterisk CLI* *astrisks*CLI dahdi show status* Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) * output when i dialing to a local number* Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-003a' -- Executing [h@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote: Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it ! On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote: Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before connecting to the FXO card, i am getting a ringing. But i get a message like the number is out of order when i just connect the line to FXO card. Please some one help me to resolve his issue -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list
Re: [asterisk-users] PSTN connectivity
Can you please figure out the configuration issue in my freepbx ? On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind govoi...@gmail.com wrote: The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI. there is some misconfiguration in FreePBX and your dialled number is not hitting any dial-able rule. See your FreePBX guide. On Thu, Sep 29, 2011 at 11:01 AM, michael k mich...@inapp.com wrote: Hi, Please see the sample. A ) Analog HardwareType Ports Action FXO Ports 1 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo FXS Ports -- B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog * * C ) ZAP Trunk (DAHDI compatibility Mode)* Trunk Description: Outbound Caller ID:CID Options: Maximum Channels: Disable Trunk: Disable Monitor Trunk Failures: Enable Outgoing Dial Rules Dial Rules: 0471+NXX Dial Rules Wizards: Outbound Dial Prefix:Outgoing Settings Zap Identifier (trunk name): *D ) INBOUND route * Description: Extensions: 199 * E ) **OUTBOUND Route* Route Name: 9_outside Route CID: Override Extension CID Route Password: PIN Set: Emergency Dialing: Intra Company Route: Music On Hold? Dial Patterns 8|NXXNXX 8|NXX Dial patterns wizards*: * Trunk SequenceZAP/g0 0 * F ) In command Line I can see the following things * [root@astrisks ~]# *dahdi_cfg -vv* DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to none [root@astrisks ~]# *dahdi_scan* [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO *Asterisk CLI* *astrisks*CLI dahdi show status* Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) * output when i dialing to a local number* Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-003a' -- Executing [h@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote: Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it ! On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote: Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before connecting to the FXO card, i am getting a ringing. But i get a message like the number is out of order when i just connect the line to FXO card. Please some one help me to resolve his issue -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk
Re: [asterisk-users] PSTN connectivity
Thanks for the update. but how do i resolve this issue ? can you help me please ? On Thu, Sep 29, 2011 at 1:00 PM, Sam Govind govoi...@gmail.com wrote: Actually its easier. I haven't worked on FreePBX lately so what I remember is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep it empty as well. Then you've created an outbound route its dial-rule is important. But the funny thing which I didn't mention before is that you've ZAP defined in FreePBX but actually its DAHDI so I remember they've this cute parameter in amportal.conf which tells FreePBX to convert ZAP into DAHDI. On Thu, Sep 29, 2011 at 11:57 AM, michael k mich...@inapp.com wrote: Can you please figure out the configuration issue in my freepbx ? On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind govoi...@gmail.com wrote: The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI. there is some misconfiguration in FreePBX and your dialled number is not hitting any dial-able rule. See your FreePBX guide. On Thu, Sep 29, 2011 at 11:01 AM, michael k mich...@inapp.com wrote: Hi, Please see the sample. A ) Analog HardwareType Ports Action FXO Ports 1 Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo FXS Ports -- B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: *from-analog * * C ) ZAP Trunk (DAHDI compatibility Mode)* Trunk Description: Outbound Caller ID:CID Options: Maximum Channels: Disable Trunk: Disable Monitor Trunk Failures: Enable Outgoing Dial Rules Dial Rules: 0471+NXX Dial Rules Wizards: Outbound Dial Prefix:Outgoing Settings Zap Identifier (trunk name): *D ) INBOUND route * Description: Extensions: 199 * E ) **OUTBOUND Route* Route Name: 9_outside Route CID: Override Extension CID Route Password: PIN Set: Emergency Dialing: Intra Company Route: Music On Hold? Dial Patterns 8|NXXNXX 8|NXX Dial patterns wizards*: * Trunk SequenceZAP/g0 0 * F ) In command Line I can see the following things * [root@astrisks ~]# *dahdi_cfg -vv* DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to none [root@astrisks ~]# *dahdi_scan* [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO *Asterisk CLI* *astrisks*CLI dahdi show status* Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) * output when i dialing to a local number* Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-003a' -- Executing [h@from-internal:1] Macro(SIP/199-003a, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/199-003a, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/199-003a, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/199-003a, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/199-003a, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote: Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it ! On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote: Hi All, I am trying to connect my asterisk box with freepbx to PSTN
[asterisk-users] OUTBOUND and INBOUND routes
Hello All, I have a pstn line can have the local, STD and ISD capabilities. My local number is 91471-2527XXX and the region is India. I would like to use the number for all possible calls ( local, STD and ISD call facilities to Land line and mobile phones) through an FXO card configured in asterisk freepbx. Can anybody help me to create an outbound route and inbound route required in freepbx for the above requirement ? Thanks, Michael.k -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PSTN connectivity
Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before connecting to the FXO card, i am getting a ringing. But i get a message like the number is out of order when i just connect the line to FXO card. Please some one help me to resolve his issue -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium ISDN card
Hi All, I am new in asterisk. In my office we have purchased ISDN pri line with 30 channels. we have more than 60 soft phone nodes and the internal asterisk connectivity between extensions are working with soft phones. Can anybody tell me which pci or pci express digium card can be used to connect my asterisk server and the ISDN pri line with 30 channels ? Please assist me to do if possible Michael.k -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing call issue in D-link DPH-80 ip phones
Hi All, Along with my asterisks server, all incoming calls to my D-link DPH-80 ip phones are are working fine while calling from soft phones with good voice clarity. But not able to make outgoing calls from the same D-link DPH-80 ip phones to either soft phone or IP phone. What would be the reason ? Any suggestions please Regards, Michael.k -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect asterisk to normal telephone PBX
Hello All, I don't even know the relevancy of my question. Please answer me if my question have some sense. I have recently implemented an asterisk server with freepbx. I have created 100 extentions and i can make successful calls between extensions from anywhere. But my office have three different land-line numbers and three of them are terminating into an internal PBX ( normal matrix telephone PBX) with more than 60 extensions. This internal PBX is the live PBX where we can call local, STD and ISD from extensions. At present i have some practical difficulties to configure telephone lines at the end of asterisk PBX. So i am trying to connect my asterisk PBX to the normal telephone PBX. I have installed 1 port x100p FXO card in my asterisk PBX and detected by my freepbx. Then i removed my normal telephone extension cable from phone and connected to the FXO port of my asterisk PBX. Ultimately my intention is that 1) if somebody call to my normal telephone extension, that should reach to my asterisk server, and asterisk server should send this call to my asterisk extension. 2) if i am calling from my asterisk extension, call should go to the normal telephone PBX via FXO card in my asterisk server and ultimately the call should send outside via the telephone PBX. Is my approach is correct ? If it is wrong please somebody assist me to connect my asterisk PBX to normal telephone PBX. Michael.K -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect asterisk to normal telephone PBX
Thanks for the reply. I am using an analog phone in normal PBX. I have an extension called 199 in asterisk and an extension 264 in analog PBX. So how do i create an inbound or outbound routes for call between these two extentions ? On Thu, Jul 28, 2011 at 1:39 PM, Carlos M Cruz carlosmoc...@gmail.comwrote: Hi, Did you created your normal Inbound and Outbound routes in freepbx? For use with your zap channels? You'll problably have to change your routes on your pbx too... Regards, Carlos M Cruz 2011/7/28 michael k mich...@inapp.com Hello All, I don't even know the relevancy of my question. Please answer me if my question have some sense. I have recently implemented an asterisk server with freepbx. I have created 100 extentions and i can make successful calls between extensions from anywhere. But my office have three different land-line numbers and three of them are terminating into an internal PBX ( normal matrix telephone PBX) with more than 60 extensions. This internal PBX is the live PBX where we can call local, STD and ISD from extensions. At present i have some practical difficulties to configure telephone lines at the end of asterisk PBX. So i am trying to connect my asterisk PBX to the normal telephone PBX. I have installed 1 port x100p FXO card in my asterisk PBX and detected by my freepbx. Then i removed my normal telephone extension cable from phone and connected to the FXO port of my asterisk PBX. Ultimately my intention is that 1) if somebody call to my normal telephone extension, that should reach to my asterisk server, and asterisk server should send this call to my asterisk extension. 2) if i am calling from my asterisk extension, call should go to the normal telephone PBX via FXO card in my asterisk server and ultimately the call should send outside via the telephone PBX. Is my approach is correct ? If it is wrong please somebody assist me to connect my asterisk PBX to normal telephone PBX. Michael.K -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot figure out pound key in qwerty keyboard
Hi All, Thanks for the reply. I have typed Shift-3 (#) but the system keep on saying that I am sorry i did not understand your response. Any other solutions to resolve this ? On Thu, Jun 30, 2011 at 8:44 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Journo *Sent:* Thursday, June 30, 2011 10:12 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Cannot figure out pound key in qwerty keyboard ** ** I am new in Asterisk. I am using asterisks with freepbx 2.7.0.10 version. I have tried to setup voice mail by dialing *97 from my extension. The prerecorded system asking for a pond key at the end of each recording. But unfortunately i am not able to locate a pound key on my qwerty key board. I have tried shift+3 but no luck. Please someone help me to figure out the pound key in my keyboard. ** ** It's this # key. #1 . # - Shift-3 should always work #2. The Asterisk Read command should “simulate” a # press after a certain time unless it wasn’t properly coded. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael.k System Administrator InApp Information Technologies (I) Pvt Ltd www.inapp.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot figure out pound key in qwerty keyboard
Actually I am using SFLphone (softphone ) in ubuntu 11.04 Desktop for dialing. On Thu, Jun 30, 2011 at 8:59 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, June 30, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cannot figure out pound key in qwerty keyboard Hi All, Thanks for the reply. I have typed Shift-3 (#) but the system keep on saying that I am sorry i did not understand your response. Any other solutions to resolve this ? The default in many phones is to use # to indicate end of dialing and not pass that digit to the PBX. You will have to check the docs for your phone. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot figure out pound key in qwerty keyboard
Yes i can see the CLI Console. In CLI display a message User cancelled message by pressing 0 after pressing Shift-3 (#) . On Thu, Jun 30, 2011 at 9:07 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *michael k *Sent:* Thursday, June 30, 2011 10:09 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Cannot figure out pound key in qwerty keyboard ** ** All, I am new in Asterisk. I am using asterisks with freepbx 2.7.0.10 version. I have tried to setup voice mail by dialing *97 from my extension. The prerecorded system asking for a pond key at the end of each recording. But unfortunately i am not able to locate a pound key on my qwerty key board. I have tried shift+3 but no luck. Please someone help me to figure out the pound key in my keyboard. Michael.k can you view the CLI console? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael.k System Administrator InApp Information Technologies (I) Pvt Ltd www.inapp.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot figure out pound key in qwerty keyboard
The CLI output after input the passcode. After pressing pound key Shift-3 (#) I am getting CLI output as User cancelled message by pressing 0 . Any suggestions ? SIP/199-00d1 Playing 'vm-password.ulaw' (language 'en') -- SIP/199-00d1 Playing 'vm-youhave.ulaw' (language 'en') -- SIP/199-00d1 Playing 'vm-no.ulaw' (language 'en') -- SIP/199-00d1 Playing 'vm-messages.ulaw' (language 'en') -- SIP/199-00d1 Playing 'vm-opts.ulaw' (language 'en') -- SIP/199-00d1 Playing 'vm-options.ulaw' (language 'en') -- Recording the message -- SIP/199-00d1 Playing 'vm-rec-unv.ulaw' (language 'en') -- SIP/199-00d1 Playing 'beep.ulaw' (language 'en') -- x=0, open writing: /var/spool/asterisk/voicemail/default/199/unavail.tmp format: wav49, 0x78a53d8 -- x=1, open writing: /var/spool/asterisk/voicemail/default/199/unavail.tmp format: wav, 0x7917ad8 -- User cancelled message by pressing 0 -- SIP/199-00d1 Playing 'vm-sorry.ulaw' (language 'en') -- SIP/199-00d1 Playing 'vm-torerecord.ulaw' (language 'en') On Thu, Jun 30, 2011 at 9:42 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Thursday 30 Jun 2011, michael k wrote: All, I am new in Asterisk. I am using asterisks with freepbx 2.7.0.10 version. I have tried to setup voice mail by dialing *97 from my extension. The prerecorded system asking for a pond key at the end of each recording. But unfortunately i am not able to locate a pound key on my qwerty key board. I have tried shift+3 but no luck. Please someone help me to figure out the pound key in my keyboard. Michael.k By pound key I presume they mean a comment mark (#), ASCII position 35 / 0x23. Also known as a number sign, hash or square. Now a boring history lesson: In the bad old days, ASCII was a 7-bit code; and each country had a national variant, replacing certain obscure punctuation marks with accented characters and other symbols peculiar to the local language and culture. In the old British ASCII variant, position 35 was indeed a £ sign (a stylised cursive L with a horizontal bar, occupying position 163 now we are using 8 bits per character). This seems to be the source of the confusion. At any rate, I have never seen a British, American or other nationality telephone with a £ key bottom right -- though I have encountered many printers that produced a £ sign instead of a #, when given ASCII code 35. On my machine (British keyboard, Linux) I can type a £ sign by pressing shift + 3; I can also get one by pressing Alt Gr + shift + 3. On a Mac, shift + 3 and option + 3 produce £ and #, one way around or the other. I don't know about Windows. One might have naïvely expected them to choose position 36 -- the dollar sign in US ASCII -- for the pound sign in British ASCII; but British practice was to write, for example, No. 1 as opposed to #1, and in any case there might be good reasons for wanting two currency symbols -- easier to sell computers to banks if they look as though they can perform conversions between currencies? Also, the British variant would have been the obvious choice in countries such as Australia and New Zealand, where the local currency is called the dollar and given the symbol $. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael.k System Administrator InApp Information Technologies (I) Pvt Ltd www.inapp.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC - INUSE Flag
It works.
I terminated the call during the playback.
AGI debug
AGI Tx 200 result=-1 endpos=480
HUP received!
Allowing
setinuse() to get called
Thanks
Michael
On 10/6/05 12:13 AM, Darren Wiebe [EMAIL PROTECTED] wrote:
Edit astcc.agi and stick these lines in before sub load_config.
$SIG{HUP} = 'ignore_hup';
sub ignore_hup {
print STDERR \nHUP received!\n\n;
}
Darren Wiebe
[EMAIL PROTECTED]
Scott Wolfe wrote:
How do you you apply the patch?
-Scott
- Original Message - From: Nicolás Gudiño [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 05, 2005 9:31 PM
Subject: Re: [Asterisk-Users] ASTCC - INUSE Flag
On 10/5/05, Darren Wiebe [EMAIL PROTECTED] wrote:
Any developers out there that would like to look at this one? It works
fine on Asterisk CVS-v1-0-09/22/05-22:23:34 on a i686 running Linux but
it does not work on the 1.2 betas. I agree that the number should be
set aside then. I wonder what the problem is.
http://bugs.digium.com/view.php?id=5400
Seems to fix the problem... please test and give feedback.
--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] ASTCC - INUSE Flag
This is my debug with the same issue The agi terminates during the sub tell_time() and exits without calling sub setinuse() or completing the reset of the script. AGI Tx agi_request: astcc.agi AGI Tx agi_channel: Zap/49-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1128401550.162 AGI Tx agi_callerid: xx AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 3 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 33 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: xx AGI Tx agi_rdnis: unknown AGI Tx agi_context: default AGI Tx agi_extension: xx AGI Tx agi_priority: 103 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: xxx AGI Tx 0-r1*CLI AGI Rx ANSWERLI AGI Tx 200 result=0 AGI Rx GET DATA astcc-enter-card-num 6000 -- Playing 'astcc-enter-card-num' (language 'en') AGI Tx 200 result=3546 AGI Rx STREAM FILE astcc-youhave 0123456789 AGI Tx 200 result=0 endpos=4480 AGI Rx SAY NUMBER 11 0123456789 -- Playing 'digits/11' (language 'en') AGI Tx 200 result=0 AGI Rx STREAM FILE astcc-dollars 0123456789 AGI Tx 200 result=0 endpos=6720 AGI Rx STREAM FILE astcc-and 0123456789 AGI Tx 200 result=0 endpos=3680 AGI Rx SAY NUMBER 88 0123456789 -- Playing 'digits/80' (language 'en') -- Channel 0/1, span 3 got hangup request AGI Tx 200 result=-1 == Spawn extension (default, x, 103) exited non-zero on 'Zap/49-1' -- Hungup 'Zap/49-1' -Michael On 10/3/05 10:52 PM, Darren Wiebe [EMAIL PROTECTED] wrote: Can you please post the output with debug agi on ? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in my astcc-exten.conf [incoming] exten = s,1,Answer ;exten = s,2,DeadAGI(astcc.agi) exten = s,2,AGI(astcc.agi) exten = s,3,Hangup I did some Google search on this issue and saw someone else had a problem but no response. -Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] ASTCC issues
I have been testing the ASTCC and have notice that when the caller hangs up the line while the balance is being played back the sub savedata() is not being called because the asterisk terminates the AGI and the rest of the script does not get executed thus never returning: AGI Script astcc.agi completed, returning 0 This leave the inuse set to 1 and the pin can not be used. I am using the lastest CVS HEAD asterisk-perl-0.08 Any comments -Michael ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] ASTCC UPDATEproblem
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-18 02:46:15 UTC I am getting this error when the astcc.agi tries to UPDATE inuse = 0 LOG: unexpected EOF on client connection (postgres on Debian) I use another astcc.agi that UPDATEs to a different server (Postgres on OSX) and the inuse = 0 gets update properly Any ideas Michael Rodriguez ___ 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
Re: [Asterisk-Users] 1-800 number
More info On 8/17/05 3:34 AM, Christoph Eicke [EMAIL PROTECTED] wrote: Hi! I'm searching for a 1-800 number that simply plays music for a long time (3mins) and no one picks up. I've bothered the ATT lines so far when trying out my SIP-PSTN connection but then always someone answered :-) Anyone have a number? Christoph ___ 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 ___ 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
Re: [Asterisk-Users] faxes
I have tested a fax call on asterisk with success. I used an IAXy on a broadband Time Warner connection. Faxes are much more sensitive than voice calls. If you have a good internet connection, faxes should complete fine. The only downfall it is recommended that you call to verify fax transmission after every fax. -Michael On 3/25/05 10:59 PM, AS [EMAIL PROTECTED] wrote: Is it possible and if so for a workstation user to send his fax via asterisk? ___ 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 ___ 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
Re: [Asterisk-Users] Re: asterisk+radius
I agree, why run to DBs. On the other hand, I have spoken with several people asking about radius support for asterisk because they have a billing solution that uses data from the radius servers to populate their billing DB. -Michael On 3/17/05 11:00 AM, Matthew Boehm [EMAIL PROTECTED] wrote: Kamran Ahmad wrote: i have written app for billing with asterisk. what is the problem in using radius. kamran Its a pain and redundant. Why run two seperate databases when 1 will do what you need? There is no native radius support for Asterisk. There is an addon, (search the wiki) but the last I heard of it, it was unstable. -Matthew ___ 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 ___ 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
[Asterisk-Users] PRI info digits question
Does anyone know how does asterisk handles INFO digit from a PRI line? Can info digit be used in extensions.conf to signal a call from a public phone? ___ 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
Re: [Asterisk-Users] voicemail cuts off / hangs up
If I am not mistaken, I believe the dial command is omitted if you do not have a sound card configured on your system (loaded module). -michael On 12/2/04 1:07 AM, Matt Hess [EMAIL PROTECTED] wrote: Does cvs tag v1-0 not have a dial command? I do not seem to have one.. dial No such command 'dial' (type 'help' for help) Henry Devito wrote: Ok try this Login into console Set verbose 15 Dial (extension of VoiceMailMain app) Dial mailbox number Dial password Hangup Does it still die? See my example below asterisk*CLI dial 777 -- Executing VoiceMailMain(OSS/dsp, ) in new stack Console call has been answered -- Playing 'vm-login' (language 'en') asterisk*CLI dial 500 -- Playing 'vm-password' (language 'en') asterisk*CLI dial 1234 -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'vm-INBOX' (language 'en') -- Playing 'vm-messages' (language 'en') -- Playing 'vm-onefor' (language 'en') -- Playing 'vm-INBOX' (language 'en') -- Playing 'vm-messages' (language 'en') asterisk*CLI hangup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up Test completed successfully.. test dialplan: exten = 555,1,Answer exten = 555,2,Wait(2) exten = 555,3,Playback(digits/0) exten = 555,4,Playback(digits/1) exten = 555,5,Playback(digits/2) exten = 555,6,Playback(digits/3) exten = 555,7,Playback(digits/4) exten = 555,8,Playback(digits/5) exten = 555,9,Playback(digits/6) exten = 555,10,Playback(digits/7) exten = 555,11,Playback(digits/8) exten = 555,12,Playback(digits/9) exten = 555,13,Busy log: -- Executing Answer(SIP/3036284315-31b3, ) in new stack -- Executing Wait(SIP/3036284315-31b3, 2) in new stack -- Executing Playback(SIP/3036284315-31b3, digits/0) in new stack -- Playing 'digits/0' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/1) in new stack -- Playing 'digits/1' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/2) in new stack -- Playing 'digits/2' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/3) in new stack -- Playing 'digits/3' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/4) in new stack -- Playing 'digits/4' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/5) in new stack -- Playing 'digits/5' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/6) in new stack -- Playing 'digits/6' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/7) in new stack -- Playing 'digits/7' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/8) in new stack -- Playing 'digits/8' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/9) in new stack -- Playing 'digits/9' (language 'en') -- Executing Busy(SIP/3036284315-31b3, ) in new stack == Spawn extension (lwn, 555, 13) exited non-zero on 'SIP/3036284315-31b3' Henry Devito wrote: Try to play a number sound file by using the Playback application, I think the voicemail uses the same app to play the digits. See if that works. exten = 500,1,Playback(digits/3) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up yup.. that's something I thought of as well.. and it's all there.. funny thing is.. I can start asterisk.. login just fine to voice mail.. I try again right away and I get that error that I had sent earlier and get cutoff.. Henry Devito wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 11:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voicemail cuts off / hangs up I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. [*] Ok Did you check to see if the /var/lib/asterisk/sounds/digits/ directory is intact? I had an install at an earlier date from the CVS that did not download all of the sounds. Just a thought. ___ Asterisk-Users mailing list
RE: [Asterisk-Users] H.323 - NO AUDIO IN BOTH DIRECTIONS
FYI I am experiencing the same problem. I have complied asterisk from the latest CVS The call connects with no audio or DTMF to either end. I tested with ulaw and g729 with no success. -Michael On Fri, 2004-06-25 at 10:55, Scott Stingel wrote: Just checking that you have installed the proper versions of both OpenH323 and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt asterisk after those installations as specified? If so, then you are having the same problem I'm experiencing: no audio on H.323. I'm also connecting through a Cisco 5300. I'm just generating audio in one direction: outbound from asterisk - I hear nothing. This used to work I'm pretty sure... There is an outstanding bug report covering H.323 problems (#1334), not sure what the current status is. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Friday, June 25, 2004 7:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Michael K. Rodriguez Dialmex LLC Director of Network Operations 200 S. 10th Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 682-8521 fax (956) 239-0627 mobile
[Asterisk-Users] Registration Error
I am using a 7960 and it is registered to the *server, but I keep getting this error. Does anyone know why? NOTICE[5126]: File chan_sip.c, Line 3080 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '67.98.37.220' Michael K. Rodriguez DialMex LLC NOC Engineer 200 S. 10th Street Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 239-0627 mobile (956) 682-5821 fax [EMAIL PROTECTED] Escalation Procedure +++The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and destroy any copies of this document.+++
[Asterisk-Users] Asterisk to gateway
Is it possible to send a call from the asterisk server to a gateway via sipv2 protocol. I have some 7960 phones that can receive a call from a 5350 via sipv2 and the phone can send to the gateway via sipv2. Is there an exten that dials to a gateways ? Michael K. Rodriguez DialMex LLC NOC Engineer 200 S. 10th Street Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 239-0627 mobile (956) 682-5821 fax [EMAIL PROTECTED] Escalation Procedure +++The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and destroy any copies of this document.+++
[Asterisk-Users] Dialout Zap1/1
Any ideas on how to dialout exten = zap 1/1 Michael K. Rodriguez DialMex LLC NOC Engineer 200 S. 10th Street Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 239-0627 mobile (956) 682-5821 fax [EMAIL PROTECTED] Escalation Procedure +++The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and destroy any copies of this document.+++
RE: [Asterisk-Users] Dialout Zap1/1
I would like to dialout the line attached to zap/1. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, March 26, 2003 1:19 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dialout Zap1/1 On Wednesday 26 March 2003 12:55 pm, Michael K. Rodriguez wrote: Any ideas on how to dialout exten = zap 1/1 Do you want to Dial the station at Zap/1? Or do you want to dial out on the telephone line attached to Zap/1? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
