[asterisk-users] unsuscribe
unsuscribe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Default mohmp3 : free of rights ?
Hello Asterisk's Users ! Is anybody knows if the default MP3 tracks provided with the lastest release of asterisk is free of rights or not ? The default tracks are : - fpm-calm-river.mp3 - fpm-sunshine.mp3 - fpm-world-mix.mp3 Regards, -- Sébastien Mortier AbsysTech Tel : +33 892 460 991 Fax : +33 320 745 005 Gsm : +33 620 792 429 Assistante : Sarah Foucart [EMAIL PROTECTED] - http://www.absystech.fr Visitez le programme d'incentive AbsysTech : http://incentive.absystech.fr ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Periodic announcements MySQL Realtime
Hi everybody, I'm trying to use the periodic-annouce and periodic-announce-frequency options. I use Asterisk 1.2.9.1-BRIstuffed-0.3.0 with MySQL realtime configuration. It seems that Asterisk Realtime queues doesn't support these options. When I try to add the 2 fields to the MySQL table, nothing happens ! Have you got any idea to solve this problem or static configuration is the only way for using these options ? Thank you by advance, Seb. ___ --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
[Asterisk-Users] warnings during parsing of misdn.conf
Hello, I have a strange problem with misdn.conf When I run asterisk, I have this message : Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config: misdn.conf: use_callingpres=yes (section: general) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config: misdn.conf: presentation=allowed (section: general) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config: misdn.conf: diaplan=0 (section: general) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. I don't understand because the parameters : - use_callingpres - presentation - dialplan are set in the correct section.. My misdn.conf : [general] debug=4 append_digits2exten=yes stop_tone_after_first_digit=yes bridging=yes use_callingpres=yes presentation=allowed diaplan=0 [default] context=default language=fr nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 [TEports] context=isdn-incoming method=round_robin ports=2,7,8 callgroup=1 msns=5500,5501,5502,5503,5504,1812,1813,1814,1815,1816,5606,5607,5608 [NTports] context=pbx-incoming method=round_robin ports=3,4,6 callgroup=2 Have you got any idea ? Thank you for your help. Seb, ___ --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] ISDN Protocol Unknom Error with Junghanns OctoBRI
Hello Henk, 1013 was the number that I tried to call. In France, it's the number to call the operator France Telecom. I've also tried to call another number without success. I will make some tests this evening with the answers that I received via this mailing-list. Thank you. Henk Dick a écrit : I think that you are sending an outgoing caller id that is not part of the DID range. Most operators do not allow this. ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ] Are you using caller id 1013 ? Change it to a number that is part of your trunks. Henk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sébastien Mortier Sent: maandag 20 maart 2006 11:52 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM -- OctoBRI -- Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h -- OctoBRI -- PABX e-Generis ISDN Phones | | SIP Phones France Telecom -- SIP Phones : Works France Telecom -- ISDN Phones : Works SIP Phones -- ISDN Phones : Works ISDN Phones - SIP Phones : Works SIP Phones -- France Telecom : DOESN'T WORK ISDN Phones - France Telecom : DOESN'T WORK Here are some characteristics of my Asterisk Setup OS Linux Gentoo 2.6.15-r1 zaptel 1.2.3 libpri 1.2.2 asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h ISDN Lines : EuroISDN not EuroISDN+ Junghanns OctoBRI PCI ISDN Card S/T 1+8 - S/T 2+7 : TE Mode S/T 3+6 - S/T 4+5 : NT Mode modprobe qozap ports=60 zaptel.conf --- loadzone=fr defaultzone=fr # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,1,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 --- zapata.conf --- switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres = yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 callprogress=yes context=isdn-incoming group = 1 ; S/T port 1,2,7,8 channel = 1-2 channel = 4-5 ;channel = 19-20 channel = 22-23 context=pbx-incoming group = 2 channel = 7-8 channel = 10-11 ;channel = 13-14 channel = 16-17 - Here's the output BRI debug when I try to make outbound calls from a SIP phone : -- Executing Dial(SIP/400-c8dc, Zap/1/1013) 1 -- Making new call for cr 137 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (Cool len=26 1 Call Ref: len= 1 (reference 9/0x9) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 811 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 051 411 801 341 301 301 ] 1 Calling Number (len= 7) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1 Presentation: Presentation permitted, user number not screened (0) '400' ] 1 [1 701 051 c11 311 301 311 331 ] 1 Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ] -- Called 1/1013 1 Protocol Discriminator: Q.931 (Cool len=8 1 Call Ref: len= 1 (reference 137/0x89) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 871 e41 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: International network (7) 1 Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] 1 -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup, cause 100 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/400-c8dc, ) == Spawn extension (default, 1013, 2) exited non-zero on 'SIP/400-c8dc' 1 received TEI check request for TEI = 127 I've already tested several configurations for zapata.conf especially with the pridialplan and switchtype lines but without success. Could you help me to analyse and solve this odd problem ? Thank you in advance, -- Sébastien Mortier AbsysTech Tel : +33 3 20 50 99 02 Fax : +33 3 20 74 50 05 Gsm : +33 6 20 79 24 29 http://www.absystech.fr
[Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI
Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM -- OctoBRI -- Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h -- OctoBRI -- PABX e-Generis ISDN Phones | | SIP Phones France Telecom -- SIP Phones : Works France Telecom -- ISDN Phones : Works SIP Phones -- ISDN Phones : Works ISDN Phones - SIP Phones : Works SIP Phones -- France Telecom : DOESN'T WORK ISDN Phones - France Telecom : DOESN'T WORK Here are some characteristics of my Asterisk Setup OS Linux Gentoo 2.6.15-r1 zaptel 1.2.3 libpri 1.2.2 asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h ISDN Lines : EuroISDN not EuroISDN+ Junghanns OctoBRI PCI ISDN Card S/T 1+8 - S/T 2+7 : TE Mode S/T 3+6 - S/T 4+5 : NT Mode modprobe qozap ports=60 zaptel.conf --- loadzone=fr defaultzone=fr # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,1,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 --- zapata.conf --- switchtype = euroisdn pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres = yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 callprogress=yes context=isdn-incoming group = 1 ; S/T port 1,2,7,8 channel = 1-2 channel = 4-5 ;channel = 19-20 channel = 22-23 context=pbx-incoming group = 2 channel = 7-8 channel = 10-11 ;channel = 13-14 channel = 16-17 - Here's the output BRI debug when I try to make outbound calls from a SIP phone : -- Executing Dial(SIP/400-c8dc, Zap/1/1013) 1 -- Making new call for cr 137 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (Cool len=26 1 Call Ref: len= 1 (reference 9/0x9) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 811 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 051 411 801 341 301 301 ] 1 Calling Number (len= 7) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1 Presentation: Presentation permitted, user number not screened (0) '400' ] 1 [1 701 051 c11 311 301 311 331 ] 1 Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ] -- Called 1/1013 1 Protocol Discriminator: Q.931 (Cool len=8 1 Call Ref: len= 1 (reference 137/0x89) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 871 e41 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: International network (7) 1 Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] 1 -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup, cause 100 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/400-c8dc, ) == Spawn extension (default, 1013, 2) exited non-zero on 'SIP/400-c8dc' 1 received TEI check request for TEI = 127 I've already tested several configurations for zapata.conf especially with the pridialplan and switchtype lines but without success. Could you help me to analyse and solve this odd problem ? Thank you in advance, -- Sébastien Mortier AbsysTech Tel : +33 3 20 50 99 02 Fax : +33 3 20 74 50 05 Gsm : +33 6 20 79 24 29 http://www.absystech.fr ___ --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
