[asterisk-users] Sound files
Hello, Can i identify the sound files that are played in the asterisk console? I defined the verbose to 100 but i can not see the sound files that are played in some situations... :( For example, I need to know what files are played for the message: Extension xxx is unavailable The goal is to translate that to Portuguese (pt_pt)... Thanks in advance, PS. ___ --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] Freepbx changes dont reflect in asterisk
Alex was right. The problem is that when i make changes in freepbx, those changes are not written in the config files. I only made modifications in files_custom.conf. The version of freePbx that i use is 2.1.1 (not beta) and Asterisk 1.2.12.1. Thanks by your help, Ps. 2006/11/18, Alex Robar [EMAIL PROTECTED]: I think you guys are all misunderstanding the problem here. Unless I'm misunderstanding, Pedro's issue is that when he makes changes in FreePBX, those changes are not written out to the config files. Alex On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: You can't do any modifications in extensions_additional.conf and sip_additional.conf. Same is true for extensions.conf and sip.conf, and other original trixbox files. As soon as you press the red bar, they are returned to their original state. For modifications, you create your own files or use sip_customs.conf and extensions_custom.conf. Please don't mix trixbox with asterisk just because its based on asterisk. Its a completely customized solution of various software packages configured to make it work according to its own requirements. For help, post on trixbox.org forums. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx changes dont reflect in asterisk
I also restarted the box and the problem is not solved :( PS 2006/11/18, Dumpolid Exeplish [EMAIL PROTECTED]: i also used to have this problem, for instance we use the pin code functionality of FreePBX and whenever i add or modify a pin number, it is not effected or changed in the config files. i dont know what causes this error but i have noticed that restarting FreePBX or re-installing the application stops this. Just restart the box On 11/18/06, Pedro Silva [EMAIL PROTECTED] wrote: Alex was right. The problem is that when i make changes in freepbx, those changes are not written in the config files. I only made modifications in files_custom.conf. The version of freePbx that i use is 2.1.1 (not beta) and Asterisk 1.2.12.1. Thanks by your help, Ps. 2006/11/18, Alex Robar [EMAIL PROTECTED]: I think you guys are all misunderstanding the problem here. Unless I'm misunderstanding, Pedro's issue is that when he makes changes in FreePBX, those changes are not written out to the config files. Alex On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: You can't do any modifications in extensions_additional.conf and sip_additional.conf. Same is true for extensions.conf and sip.conf, and other original trixbox files. As soon as you press the red bar, they are returned to their original state. For modifications, you create your own files or use sip_customs.conf and extensions_custom.conf. Please don't mix trixbox with asterisk just because its based on asterisk. Its a completely customized solution of various software packages configured to make it work according to its own requirements. For help, post on trixbox.org forums. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Freepbx changes dont reflect in asterisk
Hello, From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS. ___ --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] Freepbx changes dont reflect in asterisk
Hi, 2006/11/17, Alex Robar [EMAIL PROTECTED]: Hi Pedro, Did you press the red bar at the top of the page? Until you do this, the config files are not written out. Yes, i press the red bar and freepbx dont return any error. For example, If i add a new extension, the files extensions_addicional.conf and sip_addicional.con are supposed to be updated and are not. Best regards, PS. Alex On 11/17/06, Pedro Silva [EMAIL PROTECTED] wrote: Hello, From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] jpeglib
Hello, When i try to install the sfftobmp3.1, the tribbox box give me the following error: ... checking for TIFFOpen in -ltiff... yes checking jpeglib.h usability... no checking jpeglib.h presence... no checking for jpeglib.h... no configure: error: jpeglib.h not found I try to find packages with jpeglib but i cannot find that... :( Someone can tell me where i can find that package? Thanks in advance! PS. ___ --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] capiAnswerFax
Hello,
Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
it give me errors...
Also i tried define one extension for fax receptions but this dont works:
exten = 1,1,Goto(handle_fax,s,1)
exten = fax,1,Goto(handle_fax,s,1)
[handle_fax]
exten = s,1,capiAnswerFax(/tmp/${UNIQUEID})
exten = s,2,Hangup()
With this code, a fax call to DID 1 must be attended and the fax
stored in /tmp, right?
This not works... :(
Thanks for any kind of possible help...
PS.
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Re: [asterisk-users] capiAnswerFax
Excellent, Michiel! This works :)
You know what kind of file it is created (SFF)?
Can you send to me the example faxreceive.php?
Thanks and best regards!
PS.
2006/11/7, Michiel van Baak [EMAIL PROTECTED]:
On 15:03, Tue 07 Nov 06, Pedro Silva wrote:
Hello,
Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
it give me errors...
Also i tried define one extension for fax receptions but this dont works:
exten = 1,1,Goto(handle_fax,s,1)
exten = fax,1,Goto(handle_fax,s,1)
[handle_fax]
exten = s,1,capiAnswerFax(/tmp/${UNIQUEID})
exten = s,2,Hangup()
With this code, a fax call to DID 1 must be attended and the fax
stored in /tmp, right?
This not works... :(
Thanks for any kind of possible help...
PS.
Hi,
The chan_capi you mention already has fax support.
Here is the handle_fax context I use with the latest
released chan_capi-cm
[handle_fax]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID})
exten = s,n,capicommand(receivefax|${FAXFILE})
exten = h,1,DeadAgi(faxreceive.php|${FAXFILE})
Good luck
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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Re: [asterisk-users] Diva server 4bri and Portuguese BRI
2006/11/1, Armin Schindler [EMAIL PROTECTED]: On Wed, 1 Nov 2006, Pedro Silva wrote: As you can see in the log below, the called number is just '0': CalledPartyNumber = 810 It seems DDI 0 of your line was called. So just do exten = 0,n,Dial... Armin Is that right! Thanks! Best regards, PS. ___ --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] How to clear trixbox configuration
Hello all, To test some configs i forgot the trixbox web config (freepbx) and i made changes directly in asterisk config files (sip.conf, extensions.conf, etc). Result: asterisk is working ok but the the web config is totaly confused and, if i made a change via freepbx this not works ok. Only now i know that this changes will be made in file_custom.conf... problem of newbie... :). I also updated the asterisk for version 1.2.12.1, independently for the trixbox updating system. My trixbox version is 1.2.2. So i need to clear all configuration and start again only with the web config in freepbx. Is possible to clear all web configs and restitute all initial /etc/asterisk/* files to start from zero without re-installing all trixbox box from CD? Thanks in advance! Best regards, PS. ___ --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] Diva server 4bri and Portuguese BRI
Hello, The problem was wrong contexts defined like Marco said, and is solved. Now, i have another problem...of course :) On incoming calls, i only can receive calls if i define a line like the following, in extensions.conf: exten = _.,n,Dial(SIP/500,30,tr) (all incoming calls are redirected to extension 500). The problem is that i have some DDI's assigned by my telco (xxx302500 to xxx302509) and i need to route each DDI to diferent internal extension. If i define someting like exten = _0,n,Dial... (for DDI xxx302500) the call is not answered by asterisk. I think that asterisk cannot identify the destination DDI of the incoming call...is this normal? This is the capi debug of one incoming call: asterisk1*CLI CONNECT_IND ID=001 #0x1975 LEN=0045 Controller/PLCI/NCCI= 0x401 CIPValue= 0x10 CalledPartyNumber = 810 CallingPartyNumber = 00 83X CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default -- CONNECT_IND (PLCI=0x401,DID=0,CID=X,CIP=0x10,CONTROLLER=0x1) ISDN1#02: msn='*' DNID='0' MSN == ISDN1#02: setting format alaw - 0x8 (alaw) == ISDN1#02: Incoming call 'X' - '0' INFO_IND ID=001 #0x1976 LEN=0017 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x70 InfoElement = 810 INFO_RESP ID=001 #0x1976 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element CALLED PARTY NUMBER ISDN1#02: INFO_IND DID digits not used in this state. INFO_IND ID=001 #0x1977 LEN=0015 Controller/PLCI/NCCI= 0x401 InfoNumber = 0xa1 InfoElement = default INFO_RESP ID=001 #0x1977 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element Sending Complete CONNECT_RESP ID=001 #0x1977 LEN=0032 Controller/PLCI/NCCI= 0x401 Reject = 0x1 BProtocol B1protocol = 0x0 B2protocol = 0x0 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default ConnectedNumber = default ConnectedSubaddress = default LLC = default AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default INFO_IND ID=001 #0x1978 LEN=0016 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x18 InfoElement = 81 INFO_RESP ID=001 #0x1978 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element CHANNEL IDENTIFICATION 81 INFO_IND ID=001 #0x1979 LEN=0015 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x8005 InfoElement = default INFO_RESP ID=001 #0x1979 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element SETUP ISDN1#02: IE SETUP / SENDING-COMPLETE already received. DISCONNECT_IND ID=001 #0x197b LEN=0014 Controller/PLCI/NCCI= 0x401 Reason = 0x0 DISCONNECT_RESP ID=001 #0x197b LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: DISCONNECT_IND on incoming without pbx, doing hangup. CAPI/ISDN1/0-15: set channel task to 1 == ISDN1#02: CAPI Hangingup for PLCI=0x401 in state 4 == ISDN1#02: Interface cleanup PLCI=0x401 CAPI devicestate requested for ISDN1/0 Anyone can give me ideas about this problem? Thanks in advance! Best regards, PS. ___ --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] Diva server 4bri and Portuguese BRI
Hello, One problem is solved and another appears... :( I cannot receive incoming calls on trixbox. I defined one incoming route (any DID/any CID) and forwading these calls to a SIP extension. With capi and sip debug in asterisk -r console i dont detect any incoming activity... In xlog console i have the following debug: 0:1898:127 - SIG-R(034) 08 01 0D 05 A1 04 03 80 90 A3 18 01 81 6C 0B 00 83 39 36 33 30 34 35 37 32 33 70 02 81 30 7D 02 91 81 Q.931 CR0d SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 81 Calling Party Number 00 83 '963045723' Called Party Number 81 '0' HLC 91 81 0:1898:127 - SIG-S 0-6 e:805 0:1898:130 - CREATEID ok: context:1f assigned Id:3 freeIds=ec 0:1898:130 - alloc cr in use =4 0:1898:133 - SIG-X(008) 08 01 8D 45 08 02 80 95 Q.931 CR8d DISC Cause 80 95 'Call rejected' 0:1898:133 - SIG-x(008) 08 01 8D 5A 08 02 80 D8 Q.931 CR8d REL_COM Cause 80 d8 'Incompatible destination' 0:1898:133 - SIG-S 6-0 e:8c5 0:1898:134 - D-X(012) 00 01 14 16 08 01 8D 5A 08 02 80 D8 0:1898:135 - free cr in use =3 0:1898:135 - DELETEID ok: deleted Id:4 freeIds=ec 0:1898:155 - D-R(004) 00 01 01 16 So the problem appears to be Incompatible destination... but is problem in asterisk or is before asterisk, on diva card...? Tanks by any possible help! Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Finally this works!!! :) Tanks to Alberto and Marco by your help! The problems are: - the cable was connected to the wong card port... :( - the card config needs to be: ETSI; TE; Point-to-Point (I thought that was point-to-multipoint). Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Hello again Alberto! Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.com) This is the xlog output: 4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:074 - alloc cr in use =4 4:1736:076 - free cr in use =3 4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:078 - alloc cr in use =4 4:1736:080 - free cr in use =3 4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:081 - alloc cr in use =4 4:1736:083 - [1,0] dsp_assign 0016, 0, 2048 4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08 4:1736:084 - [1,0] Download 532 requested 4:1736:084 - MORE 4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E 02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33 Q.931 CR36 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 83 Progress Indicator 80 83 Called Party Number 80 '963045723' 4:1736:085 - SIG-S 0-1 e:885 4:1736:087 - ACTIVATION_REQ 4:1744:147 - L1_DOWN 4:1744:150 - SIG-EVENT 08 4:1744:150 - SIG-EVENT 08 4:1744:150 - EVENT: Call failed in State 'Call initiated' Link disconnected, Layer-1 error (cable or NT) 4:1744:150 - SIG-S 1-0 e: 4:1744:151 - [1,0] dsp_release 4:1744:155 - free cr in use =3 4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb I disconnect the rj45 cable from alcatel pbx and connect that to the diva card (with alcatel pbx i can make calls normally). The green led of the diva card is activated when i connect the cable. So i dont understand why the message Link disconnected, Layer-1 error (cable or NT)... This debug is th same if the cable is connected to the NT or not. Any ideas...? Thanks! PS. ___ --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] Diva server 4bri and Portuguese BRI
Thanks Alberto! I tested with telsampl like you said (with various configurations for de diva) and this not works...:( The trace is: Enter destination address: 273xx --Conn_Req(273xx) Connect_Con-- [29]:Disc_Ind-- --Disc_Res **Call cleared*** Any idea for the possible problem? Thanks and best regards, PS. 2006/10/29, Alberto Pastore [EMAIL PROTECTED]: Pedro Silva ha scritto: Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --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 I'm not sure about Portuguese operators standard, but I bet ETSI-DSS1 should work just fine. The interface mode is surely TE. The DID/MSN should not affect outgoing calls, I generally leave DID off unless the telco company has that service active. If you're using the diva server for linux package from eicon (divas4linux, currently rel. 8.2), you should find a very simple utility named telsampl under /usr/lib/eicon/divas which you can run besides asterisk, to test outgoing calls. You should run it with this command line: telsampl -c x where x is the bri port you wish to test (1..4) then at the prompt type c and enter a pstn number, e.g. your mobile phone, then you can watch the log onscreen. If the outgoing call works, then your isdn setup is correct, and the problem is in asterisk. The message from xlite is not meaningful, as it could occur on many situations. You should watch the debug output on asterisk console. That helped me a lot. Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva server 4bri and Portuguese BRI
Olá Marco! :) 2006/10/29, Marco Mouta [EMAIL PROTECTED]: pls post your misdn.conf as well as extensions.conf The asterisk version that i used with trixbox dont't have misdn.conf... I used capi.conf. For now, i dont care about asterisk, because with the divas utility telsampl i know that the problem is between diva card and BRI access. So i need to solve first this problem and only after that im care with asterisk... :) Obrigado desde já pela disponibilidade de ajuda! PS. May be i can help. Sou Português:) On 10/29/06, Pedro Silva [EMAIL PROTECTED] wrote: Thanks Alberto! I tested with telsampl like you said (with various configurations for de diva) and this not works...:( The trace is: Enter destination address: 273xx --Conn_Req(273xx) Connect_Con-- [29]:Disc_Ind-- --Disc_Res **Call cleared*** Any idea for the possible problem? Thanks and best regards, PS. 2006/10/29, Alberto Pastore [EMAIL PROTECTED]: Pedro Silva ha scritto: Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --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 I'm not sure about Portuguese operators standard, but I bet ETSI-DSS1 should work just fine. The interface mode is surely TE. The DID/MSN should not affect outgoing calls, I generally leave DID off unless the telco company has that service active. If you're using the diva server for linux package from eicon (divas4linux, currently rel. 8.2), you should find a very simple utility named telsampl under /usr/lib/eicon/divas which you can run besides asterisk, to test outgoing calls. You should run it with this command line: telsampl -c x where x is the bri port you wish to test (1..4) then at the prompt type c and enter a pstn number, e.g. your mobile phone, then you can watch the log onscreen. If the outgoing call works, then your isdn setup is correct, and the problem is in asterisk. The message from xlite is not meaningful, as it could occur on many situations. You should watch the debug output on asterisk console. That helped me a lot. Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva server 4bri and Portuguese BRI
Hello again Alberto! Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.com) This is the xlog output: 4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:074 - alloc cr in use =4 4:1736:076 - free cr in use =3 4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:078 - alloc cr in use =4 4:1736:080 - free cr in use =3 4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:081 - alloc cr in use =4 4:1736:083 - [1,0] dsp_assign 0016, 0, 2048 4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08 4:1736:084 - [1,0] Download 532 requested 4:1736:084 - MORE 4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E 02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33 Q.931 CR36 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 83 Progress Indicator 80 83 Called Party Number 80 '963045723' 4:1736:085 - SIG-S 0-1 e:885 4:1736:087 - ACTIVATION_REQ 4:1744:147 - L1_DOWN 4:1744:150 - SIG-EVENT 08 4:1744:150 - SIG-EVENT 08 4:1744:150 - EVENT: Call failed in State 'Call initiated' Link disconnected, Layer-1 error (cable or NT) 4:1744:150 - SIG-S 1-0 e: 4:1744:151 - [1,0] dsp_release 4:1744:155 - free cr in use =3 4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb I disconnect the rj45 cable from alcatel pbx and connect that to the diva card (with alcatel pbx i can make calls normally). The green led of the diva card is activated when i connect the cable. So i dont understand why the message Link disconnected, Layer-1 error (cable or NT)... This debug is th same if the cable is connected to the NT or not. Any ideas...? Thanks! PS. ___ --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] Diva server 4bri and Portuguese BRI
Finally this works!!! :) Tanks to Alberto and Marco by your help! The problems are: - the cable was connected to the wong card port... :( - the card config needs to be: ETSI; TE; Point-to-Point (I thought that was point-to-multipoint). Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Hello again Alberto! Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.com) This is the xlog output: 4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:074 - alloc cr in use =4 4:1736:076 - free cr in use =3 4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:078 - alloc cr in use =4 4:1736:080 - free cr in use =3 4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:081 - alloc cr in use =4 4:1736:083 - [1,0] dsp_assign 0016, 0, 2048 4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08 4:1736:084 - [1,0] Download 532 requested 4:1736:084 - MORE 4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E 02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33 Q.931 CR36 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 83 Progress Indicator 80 83 Called Party Number 80 '963045723' 4:1736:085 - SIG-S 0-1 e:885 4:1736:087 - ACTIVATION_REQ 4:1744:147 - L1_DOWN 4:1744:150 - SIG-EVENT 08 4:1744:150 - SIG-EVENT 08 4:1744:150 - EVENT: Call failed in State 'Call initiated' Link disconnected, Layer-1 error (cable or NT) 4:1744:150 - SIG-S 1-0 e: 4:1744:151 - [1,0] dsp_release 4:1744:155 - free cr in use =3 4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb I disconnect the rj45 cable from alcatel pbx and connect that to the diva card (with alcatel pbx i can make calls normally). The green led of the diva card is activated when i connect the cable. So i dont understand why the message Link disconnected, Layer-1 error (cable or NT)... This debug is th same if the cable is connected to the NT or not. Any ideas...? Thanks! PS. ___ --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] Diva server 4bri and Portuguese BRI
Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --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
