Re: [Asterisk-Users] Problem with call to IAX
I find it !!! I have a bad syntax in extension.conf Sorry - Original Message - From: Rattana BIV To: [EMAIL PROTECTED] Sent: Thursday, February 19, 2004 2:27 PM Subject: [Asterisk-Users] Problem with call to IAX Hi, I've got a problem with Call to IAX. When I call from phone (use CAPI channel) to IAX I have this to asterisk : -- Called 192.168.1.22 -- Hungup 'IAX[rbiv:5036]/4' So in iax softphone call come but immediatly hungup. I don't know why... Any suggestions ? Best Regards Rattana
[Asterisk-Users] Record communication
Hi, Just a little question ... Is there a way to Record a conversation during a communication with asterisk. Perhaps with AGI commands ? Regards Rattana
[Asterisk-Users] Record conversation
Hi, Does anybody know if it is possible to record a conversation with asterisk ? Regards Rattana
Re: [Asterisk-Users] IAX call problems
No I don't have enable jitterbuffer i will test with it. Thanks - Original Message - From: Dan Tucny [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 30, 2004 8:20 PM Subject: Re: [Asterisk-Users] IAX call problems Hi Rattana, Do you have jitterbuffer enabled? Dan On Fri, 2004-01-30 at 13:40, Rattana BIV wrote: hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphones end after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 1 [192.168.1.22:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65795ms SCall: 6 DCall: 21588 [192.168.1.22:4569] Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65795ms SCall: 6 DCall: 21588 [192.168.1.22:4569] Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 65795ms SCall: 21588 DCall: 6 [192.168.1.22:4569] Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 007 Type: IAX Subclass: PING Timestamp: 75906ms SCall: 22105 DCall: 5 [192.168.1.77:4569] Tx-Frame Retry[-01] -- OSeqno: 007 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 75906ms SCall: 5 DCall: 22105 [192.168.1.77:4569] Any suggestions ??? Thanks in advance Rattana PS: The softphone I use work with wiax.dll and is developpe by me =) ___ 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 ___ 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
Re: [Asterisk-Users] IAX call problems
I have enable it and set maxjitterbuffer=900 and maxexccessbuffer =200. But communication between IAX/IAX stop after 1'10. I try with a other softphone (IaxPhone) I have the same problem. After 1'10 no sound. Don't know where the problem... Best Regards - Original Message - From: Rattana BIV [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 02, 2004 9:37 AM Subject: Re: [Asterisk-Users] IAX call problems No I don't have enable jitterbuffer i will test with it. Thanks - Original Message - From: Dan Tucny [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 30, 2004 8:20 PM Subject: Re: [Asterisk-Users] IAX call problems Hi Rattana, Do you have jitterbuffer enabled? Dan On Fri, 2004-01-30 at 13:40, Rattana BIV wrote: hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphones end after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 1 [192.168.1.22:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65795ms SCall: 6 DCall: 21588 [192.168.1.22:4569] Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65795ms SCall: 6 DCall: 21588 [192.168.1.22:4569] Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 65795ms SCall: 21588 DCall: 6 [192.168.1.22:4569] Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 007 Type: IAX Subclass: PING Timestamp: 75906ms SCall: 22105 DCall: 5 [192.168.1.77:4569] Tx-Frame Retry[-01] -- OSeqno: 007 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 75906ms SCall: 5 DCall: 22105 [192.168.1.77:4569] Any suggestions ??? Thanks in advance Rattana PS: The softphone I use work with wiax.dll and is developpe by me =) ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] IAX call problems
With DIAX096 I have the same issue... Does it work in your case ? - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 02, 2004 11:44 AM Subject: Re: [Asterisk-Users] IAX call problems Hi, - Original Message - From: Rattana BIV [EMAIL PROTECTED] I have enable it and set maxjitterbuffer=900 and maxexccessbuffer =200. But communication between IAX/IAX stop after 1'10. I try with a other softphone (IaxPhone) I have the same problem. After 1'10 no sound. Don't know where the problem... Have you tried DIAX with both IAX and IAX2 and it is the same issue? BR, Dan ___ 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 ___ 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
[Asterisk-Users] IAX call problems
hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphonesend after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 1 [192.168.1.22:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65795ms SCall: 6 DCall: 21588 [192.168.1.22:4569] Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65795ms SCall: 6 DCall: 21588 [192.168.1.22:4569] Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 65795ms SCall: 21588 DCall: 6 [192.168.1.22:4569] Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 007 Type: IAX Subclass: PING Timestamp: 75906ms SCall: 22105 DCall: 5 [192.168.1.77:4569] Tx-Frame Retry[-01] -- OSeqno: 007 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 75906ms SCall: 5 DCall: 22105 [192.168.1.77:4569] Any suggestions ??? Thanks in advance Rattana PS: The softphone I use work with wiax.dll and is developpe by me =)
[Asterisk-Users] wiax
Hi, I try to use wiax.dll in a application. Is there some docs about this DLL ? Sample ? Regards Rattana
[Asterisk-Users] Transfert with IAX
Hi, I try to use Libiax in order to put un transfert button inmy iax softphone. Is there a way to make a call transfert ? Best regards rattana
[Asterisk-Users] IAX clients
Hi, Is there IAX client in Applet JAVA which can be embeded in a web page ? Best regards Rattana
[Asterisk-Users] SIP silence detection
Hi; Just a little question about SIP. Is there silence detection with SIP ? If yes can I suppress it ? I use asterisk with SJPhone and I think there silence detection or maybe my ear doesn't hear well :) Regards Rattana
[Asterisk-Users] Notice with asterisk System application
Hi, I notice something with asterisk with the System application. When I lauch asterisk with -c option the application System work correctly. But when I lauch asterisk without option, the application System doesn't lauch command. It is normal ? Regards Rattana
[Asterisk-Users] capi config
Hi, I have DIVA server BRI with 2 channels and i use chan_capi drivers. But I only can use 1 channel. I make one call it works, but if I make a second call asterisk says me = Everyone is busy at this time. How can I configure it ? Best regards Rattana
Re: [Asterisk-Users] SIP client
Thanks very much !! I thinks it could be very useful for me Regards Rattana - Original Message - From: Peer Oliver schmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 7:14 PM Subject: Re: [Asterisk-Users] SIP client Christopher Stephens schrieb: Is there SIP client which work with Asterisk and can be embedded in a HTML page ? It may not be *exactly* what you're looking for, but try: http://fwd.pulver.com/callme.php?userid=411 [..] Unfortunately this seem to work with Internet Explorer, only. rgds pos ___ 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
[Asterisk-Users] SIP client
hi everybody, Is there SIP client which work with Asterisk and can be embedded in a HTML page ? Thanks Rattana
[Asterisk-Users] SIP how to
Hi, I try to use asterisk with SIP with Messenger client. Does anyones have already done this ? Can we make alias like asterisk-oh323 channel drivers ? What sort of extension have I put in extensions.conf ? Regards Rattana
[Asterisk-Users] CAPI silence detection
Hi, Where can I disable silence detection with chan_capi ? Is there option in capi.conf ? Regards Rattana
Re: [Asterisk-Users] CAPI silence detection
OK When I call Netmeeting by my phone. I have silence (the sound is choppy) in my phone but not with netmeeting. And I don't know why ? How can I set it ? If chan_capi not support silence detection perhaps asterisk do it ... any tips ... Thanks Rattana - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 10:26 AM Subject: Re: [Asterisk-Users] CAPI silence detection hi, there is NO option to disable silence detection for chan_capi, because chan_capi DOES NOT support silence detection. :-) regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mit, 2003-09-10 um 10.28 schrieb Rattana BIV: Hi, Where can I disable silence detection with chan_capi ? Is there option in capi.conf ? Regards Rattana ___ 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
Re: [Asterisk-Users] CAPI silence detection
I think I have found the problem . The guilty is Netmeeting !! Netmeeting make silence detection. Whatever I put silence detection into Min I have this. I make some test with openphone and I have good result. Rattana - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 11:06 AM Subject: Re: [Asterisk-Users] CAPI silence detection do you see DATA_B3_REQ errors on the * console? if yes, then netmeeting has a bad timing (or no timing?) for sending audio or your network produces that jitter. chan_capi needs to get the outgoing audio in a rather strict timing (there is no possibility to use a jitterbuffer on a Bchannel! 64 kbits is 64 kbits, if something comes too late it will create jitter). regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mit, 2003-09-10 um 10.59 schrieb Rattana BIV: OK When I call Netmeeting by my phone. I have silence (the sound is choppy) in my phone but not with netmeeting. And I don't know why ? How can I set it ? If chan_capi not support silence detection perhaps asterisk do it ... any tips ... Thanks Rattana - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 10:26 AM Subject: Re: [Asterisk-Users] CAPI silence detection hi, there is NO option to disable silence detection for chan_capi, because chan_capi DOES NOT support silence detection. :-) regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mit, 2003-09-10 um 10.28 schrieb Rattana BIV: Hi, Where can I disable silence detection with chan_capi ? Is there option in capi.conf ? Regards Rattana ___ 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 ___ 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
[Asterisk-Users] delay problem in h323
Hi, I have a delay betweentwo H323. Netmeeting1 - | | |gnuGK | --- [asterisk-oh323] | Asterisk | Netmeeting2 --| | Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2 receive the voice without delay. But in the other way I have 3 secondes delay. In oh323.conf I set jittermin and jittermax to 20, the ipTos=lowdelay. I try to find where I can delete the delay. Does anyone have a tip ? Best Regards Rattana
[Asterisk-Users] asterisk-oh323 question
Hi; Where I can find Caller IP adress or caller Login in chan_oh323.c (asterisk-oh323-0.5.5) ? I try "cd.call_source_alias" but I don't have this. Regards Rattana
[Asterisk-Users] chan_oh323 delay
Hi, I've got a big delay between the caller and the callee. I use asterisk-oh323 drivers how can I configure oh323.conf in order to have the best delay ? Thanks Rattana
Re: [Asterisk-Users] gnuGK + h323 Caller ID
How can you have that ? In my case I have (N/A) in callerID when I do show channel (the h323 Channel) Have you some advice ? Regards Rattana - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 02, 2003 6:36 AM Subject: Re: [Asterisk-Users] gnuGK + h323 Caller ID For some reason, chan_h323 ignores the callerid and puts your IP address in instead. I've modded my chan_h323 to use the caller's id instead. Trival change but I'm guessing there's a reason why it isn't so in the first place. Anyone know why? Adam Hart - Original Message - From: Rattana BIV To: [EMAIL PROTECTED] Sent: Monday, September 01, 2003 7:19 PM Subject: [Asterisk-Users] gnuGK + h323 Caller ID Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana ___ 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
[Asterisk-Users] gnuGK + h323 Caller ID
Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana
Re: [Asterisk-Users] Change include contexts runtime
I have the same problem - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist (E-mail) [EMAIL PROTECTED] Sent: Monday, September 01, 2003 10:51 AM Subject: [Asterisk-Users] Change include contexts runtime Hi there How do I change the dialplan runtime, if I for example wants all calls on the main number to be answered by a voicemail (when it is out-of-office hours). I want to be able to change the configuration by pressing a DTMF combination e.g. *82. Can't figure out whether it is necessary to change contexts or how to do it. I have read a lot of examples and config documentation, but I can't figure out how to do it. I know there are commands from the CLI to include and not include contexts but I can't get them to work. If i write 'include context in default' I can see by 'show dialplan' that 'context' is included in default. But if I want to include a context named office by typing 'include office in default' I just get 'No such command 'include office' (type 'help for help) regards Mickey Binder ___ 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
[Asterisk-Users] H323 caller ID
Hi, I use asterisk-oh323 and a gatekeeper (gnugk) and netmeeting In asterisk i can have Caller ID when I do "show channel " I have (N/A). Does anyone know how I can have this caller ID ? Notice that in the gatekeeper I can see the user login of the netmeeting caller. Regards Rattana
[Asterisk-Users] include context
hi, how can I add or remove this line "include =context"by the command CLI ? regards Rattana
[Asterisk-Users] Alias limitation in asterisk-oh323.0.5.5
Hi, Is there limitation of alias creation in the file oh323.conf with asterisk-oh323.0.5.5 ? In my config file I have 32 alias and when i call someone in h323 Asterisk do a segmentation fault. I try to delete alias and have 9, everything is OK. It is normal ? Regards Rattana
Re: [Asterisk-Users] H323 CallerID
The Caller ID is correctly passed when I receive a call from Phones. But when Netmeeting call asterisk I only have the name of the channel like H323:26022. When I do in asterisk CLI the command : show channel H323:26022. I have next Caller ID : (N/A) I try to find a way to have IP number in this caller ID. - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 07, 2003 3:46 PM Subject: Re: [Asterisk-Users] H323 CallerID Rattana BIV wrote: I run with asterisk-oh323 0.5.4 from inaccessnetwork. What message do you get in your mailbox? asterisk-oh323 does handle correctly and passes the called ID number. Thanks Rattana Michael. ___ 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
Re: [Asterisk-Users] H323 CallerID
I run with asterisk-oh323 0.5.4 from inaccessnetwork. Thanks Rattana - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 04, 2003 5:39 PM Subject: Re: [Asterisk-Users] H323 CallerID Which H.323 channel driver are you running? I don't have any H.323 endpoints ever leaving me voicemail, but I know chan_h323 does now deal with CallerID. I cannot speak for the 3rd party driver. Jeremy McNamara Rattana BIV wrote: Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana ___ 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
Re: [Asterisk-Users] segmentation fault with asterisk and OH323
It's OK. I change my oh323.conf file and I don't have segmantation fault anymore. Thanks Rattana - Original Message - From: Rattana BIV [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 3:42 PM Subject: Re: [Asterisk-Users] segmentation fault with asterisk and OH323 By my phone I call H323 client (Netmeeting) I can talk. Everything is OK. But when I hangup I have the segmentation fault. These are the last asterisk log before the segmentation fault: -- Called 192.168.1.200 -- H323:193 answered CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2145 (ast_channel_bridge): Didn't get a frame from channel: CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2213 (ast_channel_bridge): Bridge stops bridging channels CAPI[contr1/26]/1 and H323:193 Something strange is, when I call the phone from netmeeting it's work perfectly. What do you think ? (you can see my oh323.conf in attach file) Rattana - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 3:07 PM Subject: Re: [Asterisk-Users] segmentation fault with asterisk and OH323 Rattana BIV wrote: Hi, I got a segmantation fault When I call to computer (h323) from phone. I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. More info (config files, screen log, backtrace of core file) is needed. Someone know where the problem ? Regards Rattana Michael. ___ 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
[Asterisk-Users] CDR
Hi, In the file Master.csv (var/log/asterisk/cdr-csv) we have stats of call. But Call from H323 client doesn't here. What sould have do in order to have this. Regards Rattana
[Asterisk-Users] H323 CallerID
Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana
[Asterisk-Users] segmentation fault with asterisk and OH323
Hi, I got a segmantation fault When I call to computer (h323) from phone. I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. Someone know where the problem ? Regards Rattana
Re: [Asterisk-Users] segmentation fault with asterisk and OH323
By my phone I call H323 client (Netmeeting) I can talk. Everything is OK. But when I hangup I have the segmentation fault. These are the last asterisk log before the segmentation fault: -- Called 192.168.1.200 -- H323:193 answered CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2145 (ast_channel_bridge): Didn't get a frame from channel: CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2213 (ast_channel_bridge): Bridge stops bridging channels CAPI[contr1/26]/1 and H323:193 Something strange is, when I call the phone from netmeeting it's work perfectly. What do you think ? (you can see my oh323.conf in attach file) Rattana - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 3:07 PM Subject: Re: [Asterisk-Users] segmentation fault with asterisk and OH323 Rattana BIV wrote: Hi, I got a segmantation fault When I call to computer (h323) from phone. I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. More info (config files, screen log, backtrace of core file) is needed. Someone know where the problem ? Regards Rattana Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users oh323.conf Description: Binary data
[Asterisk-Users] Manager
Hi, What can I do with Manager ? Is there some docs about it ? Rattana
[Asterisk-Users] Some stats
Hi, I try to make some statistics about call on Asterisk. Is there something who makes it ? I will be interesting to have the time of a call and a list of current calls. Regards Rattana
[Asterisk-Users] Compilations errors
Hi, I try to compil the nex cvs version of asterisk cvs and i have this error gcc -shared -Xlinker -x -o cdr_mysql.so r_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib /usr/bin/ld: cannot find -lmysqlclient collect2: ld returned 1 exit status make[1]: *** [cdr_mysql.so] Error 1 make[1]: Leaving directory `/eqcall/sources/new/asterisk/cdr' make: *** [subdirs] Error 1 What should I do ? Regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfert call
Yeah ! It is good I will try now Rattana - Original Message - From: carlos del mayor [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 08, 2003 5:22 PM Subject: Re: [Asterisk-Users] Transfert call Sorry! Didn't know it got implemented!!Last notice I had is that it would be implemented soon, but didn't think it was SO soon...Great then!! cmayor --- Martin Pycko [EMAIL PROTECTED] escribió: That got implemented recently ... Martin On Tue, 8 Jul 2003, carlos del mayor wrote: Hi Rattana, That kind of transfer is not yet implemented in *. The way it will be indicated is: exten =111,dial,Zap/1,20,T The T indicate that transfer is permitted for calling party, but as I've said, that's not implemented at the moment. Regards cmayor --- Rattana BIV [EMAIL PROTECTED] escribió: Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ 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 ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ 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
[Asterisk-Users] Matching winth asterisk-oh323
Hi, It is possible to do matching in oh323.conf with asterisk-oh323? example : alias=0XXX Regards Rattana
[Asterisk-Users] Conferences with CAPI and H323
Hi; I try to work with conferences room by MeetMe and ztdummy modules. I notice that Conference with phones work well. But conference with H323 terminals work but there no sound. And conferences between H323 and capi only CAPI can talk and hear something. Does anyone has successfully make this ? Regards Rattana
[Asterisk-Users] Transfert call
Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana
[Asterisk-Users] EICON Drivers and CAPI
Hi, Does anyone use EICON Diva Server Card ISDN and CAPI ? I need some help for drivers installation. I use RED HAT 8.0 (kernel 2.4.18-14) I want to know which drivers should I use and how can I install it. Thanks Rattana
[Asterisk-Users] Chan_capi compil error
Hi; I Use RED HAT 8 and chan_capi 0.1.0 I have errors in compilation. it can't find capi20.h and i have subscripted value is neither array nor pointer. help me please. before i have red hat 7.2 it work so i don't why there're errors. Rattana
[Asterisk-Users] Problem with ztdummy
Hi, I have some problem with ztdummy in order to use Conference in asterisk. When I do modprobe ztdummy I have this : /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol proc_mkdir_Rsmp_220b03b4 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol create_proc_entry_Rsmp_3a9bfbd2 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_output_wakeup_Rsmp_fc4c9ef0 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __write_lock_failed /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol request_module_Rsmp_27e4dc04 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol register_chrdev_Rsmp_65405f1d /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol remove_proc_entry_Rsmp_cfd23da1 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __tasklet_schedule_Rsmp_ed5c73bf /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol tasklet_kill_Rsmp_79ad224b /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_input_error_Rsmp_14a5ddc7 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __generic_copy_to_user_Rsmp_d523fdd3 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_unregister_channel_Rsmp_d0b1c903 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol unregister_chrdev_Rsmp_c192d491 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol remove_wait_queue_Rsmp_2d22232f /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol alloc_skb_Rsmp_3702adc3 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __read_lock_failed /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __wake_up_Rsmp_127fda83 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol skb_over_panic_Rsmp_7acbf56a /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_register_channel_Rsmp_bd63214e /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol printk_Rsmp_1b7d4074 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol kfree_Rsmp_037a0cba /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol tasklet_init_Rsmp_a5808bbf /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __kfree_skb_Rsmp_980007b5 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol cpu_raise_softirq_Rsmp_d01f3ee8 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_unit_number_Rsmp_960fd73e /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol sprintf_Rsmp_3c2c5af5 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_channel_index_Rsmp_dc85af48 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol kmalloc_Rsmp_93d4cfe6 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_input_Rsmp_65b5d6aa /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __pollwait_Rsmp_57ac46fa /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol schedule_Rsmp_4292364c /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __generic_copy_from_user_Rsmp_116166aa /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol softnet_data_Rsmp_20eece54 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol add_wait_queue_Rsmp_0987e9bc /lib/modules/2.4.7-10/misc/zaptel.o: insmod /lib/modules/2.4.7-10/misc/zaptel.o failed /lib/modules/2.4.7-10/misc/zaptel.o: insmod ztdummy failed Notice that I don't use Zaptel device. So I think strange that I need ztdummy in order to use conference. Does anyone have a tip ? Regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SQL
I try to work on it, but I just begin. Regards Rattana - Message d'origine - De : George Lin [EMAIL PROTECTED] À : [EMAIL PROTECTED] Envoyé : mercredi 12 mars 2003 10:41 Objet : [Asterisk-Users] SQL Hello everyone, I would like to have soneone who knows how to use SQL method to update asterisk's conf files, without disrupting the ongoing calls. Can someone give me some sample about using SQL to update conf files. Thanks George Lin ___ 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
Re: [Asterisk-Users] variable in extension.conf
I try to detect if an user who use Netmeeting is connected or not.
I think in order to do that, Netmeeting-user open a web page (in PHP) et
press the button Connect or Disconnect and the PHP set the Environnement
variable which will be proceeded in extension.conf
So i need Environnement Variable, I have test it with :
s,1,SetVar,toto=$VARENV where VARENV is my environnement variable but toto
not take the value. perhaps should I try toto=${VARENV} or toto=${$VARENV}.
Regards
Rattana
- Message d'origine -
De : Steven Critchfield [EMAIL PROTECTED]
À : [EMAIL PROTECTED]
Envoyé : lundi 10 mars 2003 18:27
Objet : Re: [Asterisk-Users] variable in extension.conf
On Mon, 2003-03-10 at 11:00, Rattana BIV wrote:
Hi,
How can we use Environnement variable in extension.conf ?
I don't think you can at this moment, and I am not sure that would be a
good idea as you asterisk thread should be long lived and therefore
anything you would put in an environment variable would just as well be
placed in a config file. Not to mention a config file can be reread
without stopping asterisk or interupting running calls.
Could you point out what you wanted to accomplish with environment
variables so we can see if there is a better way of communicating the
information to asterisk with you, or see what I am currently
overlooking.
--
Steven Critchfield [EMAIL PROTECTED]
___
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[Asterisk-Users] Conference with CAPI
Hi, Just a little question... Can we do conference with CAPI (chan_capi) ? Rattana Regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323-0.5.1
How can you use DTMF detection with netmeeting ? Do you have to set something in configuration file ? Regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a problem with MeetMe
No I use chan_capi and H323 but not zaptel device. So can I use it ? When I lauch ztdummy I have some errors. Regards Rattana - Message d'origine - De : Brancaleoni Matteo [EMAIL PROTECTED] À : [EMAIL PROTECTED] Envoyé : mardi 4 mars 2003 18:32 Objet : Re: [Asterisk-Users] a problem with MeetMe have you any zaptel device in your box? a zaptel device is required for timing source for the conference (so meetme) matteo Il mar, 2003-03-04 alle 16:49, Rattana BIV ha scritto: Hi, I try the application MeeMe but i Have a problem when I call a conference. It show me : Unable to open pseudo channel Does anyone can help me ? regards Rattana ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How sample.call is proceeded
Hi, I wanted to know in which code source the file sample.call is proceeded when we put it in /var/spool/asterisk/outgoing/ I try to make an application to asterisk who check when an user in H323 (netmeeting) is connect or not. Regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a problem with MeetMe
it is errors I have when I lauch modprobe ztdummy /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol proc_mkdir_Rsmp_220b03b4 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol create_proc_entry_Rsmp_3a9bfbd2 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_output_wakeup_Rsmp_fc4c9ef0 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __write_lock_failed /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol request_module_Rsmp_27e4dc04 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol register_chrdev_Rsmp_65405f1d /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol remove_proc_entry_Rsmp_cfd23da1 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol irq_stat_Rsmp_0792a433 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_input_error_Rsmp_14a5ddc7 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __generic_copy_to_user_Rsmp_d523fdd3 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_unregister_channel_Rsmp_d0b1c903 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol unregister_chrdev_Rsmp_c192d491 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol remove_wait_queue_Rsmp_2d22232f /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol alloc_skb_Rsmp_3702adc3 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __read_lock_failed /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __wake_up_Rsmp_127fda83 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol skb_over_panic_Rsmp_7acbf56a /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_register_channel_Rsmp_bd63214e /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol printk_Rsmp_1b7d4074 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol kfree_Rsmp_037a0cba /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __kfree_skb_Rsmp_980007b5 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol cpu_raise_softirq_Rsmp_d01f3ee8 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_unit_number_Rsmp_960fd73e /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol sprintf_Rsmp_3c2c5af5 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_channel_index_Rsmp_dc85af48 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol kmalloc_Rsmp_93d4cfe6 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol ppp_input_Rsmp_65b5d6aa /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __pollwait_Rsmp_57ac46fa /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol schedule_Rsmp_4292364c /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol __generic_copy_from_user_Rsmp_116166aa /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol softnet_data_Rsmp_20eece54 /lib/modules/2.4.7-10/misc/zaptel.o: unresolved symbol add_wait_queue_Rsmp_0987e9bc /lib/modules/2.4.7-10/misc/zaptel.o: insmod /lib/modules/2.4.7-10/misc/zaptel.o failed /lib/modules/2.4.7-10/misc/zaptel.o: insmod ztdummy failed Thanks for helps Rattana - Message d'origine - De : James Sizemore [EMAIL PROTECTED] À : [EMAIL PROTECTED] Envoyé : mercredi 5 mars 2003 14:48 Objet : Re: [Asterisk-Users] a problem with MeetMe The only problem I can think that you would have with the ztdummy would be that to used a kernel source other then the one your running when you build it... So what errors did you get when you build ztdummy? Rattana BIV wrote: No I use chan_capi and H323 but not zaptel device. So can I use it ? When I lauch ztdummy I have some errors. Regards Rattana - Message d'origine - De : Brancaleoni Matteo [EMAIL PROTECTED] À : [EMAIL PROTECTED] Envoyé : mardi 4 mars 2003 18:32 Objet : Re: [Asterisk-Users] a problem with MeetMe have you any zaptel device in your box? a zaptel device is required for timing source for the conference (so meetme) matteo Il mar, 2003-03-04 alle 16:49, Rattana BIV ha scritto: Hi, I try the application MeeMe but i Have a problem when I call a conference. It show me : Unable to open pseudo channel Does anyone can help me ? regards Rattana ___ 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 ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a problem with MeetMe
Hi, I try the application MeeMe but i Have a problem when I call a conference. It show me : Unable to open pseudo channel Does anyone can help me ? regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
