Re: [asterisk-users] asterisk not detecting hangup
I've enabled those options but it's the same. On 10/25/06, Maxi Belino [EMAIL PROTECTED] wrote: i'm having similar problems (if you find out the solution please post it) did you try enabling 'callprogress' or 'busydetect' in zapata.conf ? Maxi 2006/10/23, Arkaitz [EMAIL PROTECTED]: Hi, Im working with the following versions: -asterisk-1.2.12.1 -zaptel-1.2.9.1 And with the following card: 00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 201 I/O ports at c800 [size=256] Memory at fe00 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Identified as: *CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard X101P Board 1 OK 0 0 0 And the following lines in zapata.conf(for spanish lines): answeronpolarityswitch=yes hanguponpolarityswitch=yes The problem is that although the calls work correctly the system is unable to detect a pstn hangup and it keeps running even when the other side is calling to another number(not an asterisk ones, asterisk line keeps busy) Any hint? Thanks for your time -- Arkaitz ___ --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 -- Arkaitz ___ --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] asterisk not detecting hangup
Hi, Im working with the following versions: -asterisk-1.2.12.1 -zaptel-1.2.9.1 And with the following card: 00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 201 I/O ports at c800 [size=256] Memory at fe00 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Identified as: *CLI zap show status Description Alarms IRQbpviol CRC4 Wildcard X101P Board 1 OK 0 0 0 And the following lines in zapata.conf(for spanish lines): answeronpolarityswitch=yes hanguponpolarityswitch=yes The problem is that although the calls work correctly the system is unable to detect a pstn hangup and it keeps running even when the other side is calling to another number(not an asterisk ones, asterisk line keeps busy) Any hint? Thanks for your time -- Arkaitz ___ --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] Defining sip users through mysql
Hi all, I had the same problem before so i tried reinstalling and using all the defaults i could. But unfortunatelly it's the same, it seems that mysql defined users can't access to the codec files. In both situations the phone registers and i can see it with sip show peers(with rtcachefriends=yes for mysql). Using default context with default extensions.conf Using sip.conf: [linksys] callerid=linksys type=friend user=linksys secret=linksys context=default host=dynamic If I call extension 2: -- Executing BackGround(SIP/linksys-081a5678, demo-moreinfo) in new stack -- Playing 'demo-moreinfo' (language 'en') -- Executing Goto(SIP/linksys-081a5678, s|instruct) in new stack -- Goto (default,s,6) -- Executing BackGround(SIP/linksys-081a5678, demo-instruct) in new stack -- Playing 'demo-instruct' (language 'en') -- Executing WaitExten(SIP/linksys-081a5678, ) in new stack -- Timeout on SIP/linksys-081a5678, going to 't' -- Executing Goto(SIP/linksys-081a5678, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/linksys-081a5678, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/linksys-081a5678, ) in new stack All correct. Using mysql: mysql select callerid,type,name,secret,context,host,allow,disallow from sip; +--++-+-+-+-+-+--+ | callerid | type | name| secret | context | host| allow | disallow | +--++-+-+-+-+-+--+ | Linksys | friend | linksys | linksys | default | dynamic | g729;ilbc;gsm;ulaw;alaw | all | +--++-+-+-+-+-+--+ If I call extension 2: -- SIP Seeding peer from astdb: 'linksys' at [EMAIL PROTECTED]:5060 for 3600 -- Executing BackGround(SIP/linksys-081a5678, demo-moreinfo) in new stack Oct 3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to gsm Oct 3 13:16:17 WARNING[13896]: file.c:824 ast_streamfile: Unable to open demo-moreinfo (format g729): No such file or directory Oct 3 13:16:17 WARNING[13896]: pbx.c:5798 pbx_builtin_background: ast_streamfile failed on SIP/linksys-081a5678 for demo-moreinfo -- Executing Goto(SIP/linksys-081a5678, s|instruct) in new stack -- Goto (default,s,6) -- Executing BackGround(SIP/linksys-081a5678, demo-instruct) in new stack Oct 3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to gsm Oct 3 13:16:17 WARNING[13896]: file.c:824 ast_streamfile: Unable to open demo-instruct (format g729): No such file or directory Oct 3 13:16:17 WARNING[13896]: pbx.c:5798 pbx_builtin_background: ast_streamfile failed on SIP/linksys-081a5678 for demo-instruct -- Executing WaitExten(SIP/linksys-081a5678, ) in new stack Oct 3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to g729 Oct 3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to g729 -- Timeout on SIP/linksys-081a5678, going to 't' -- Executing Goto(SIP/linksys-081a5678, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/linksys-081a5678, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') I only hear demo-thanks. Any hint please? Thanks for your time -- Arkaitz ___ --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] Re: Defining sip users through mysql
Hi again, I've found a strange thing. If I use the same account from a softphone(twinkle) it behaves well. As fas as i've tried, connecting both phones, a linksys phone and twinkle, at the same time to the same account and calling to the same extension asterisk works well for the twinkle but fails for the linksys like this: Oct 3 17:26:30 WARNING[15685]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to gsm Oct 3 17:26:30 WARNING[15685]: file.c:824 ast_streamfile: Unable to open demo-instruct (format g729): No such file or directory Oct 3 17:26:30 WARNING[15685]: pbx.c:5798 pbx_builtin_background: ast_streamfile failed on SIP/linksys-081ac210 for demo-instruct -- Executing WaitExten(SIP/linksys-081ac210, ) in new stack Oct 3 17:26:30 WARNING[15685]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to g729 Oct 3 17:26:30 WARNING[15685]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to g729 I can't find why the file paths should be different depending on the sip client. Any hint? Thanks for your time On 10/3/06, Arkaitz [EMAIL PROTECTED] wrote: Hi all, I had the same problem before so i tried reinstalling and using all the defaults i could. But unfortunatelly it's the same, it seems that mysql defined users can't access to the codec files. In both situations the phone registers and i can see it with sip show peers(with rtcachefriends=yes for mysql). Using default context with default extensions.conf Using sip.conf: [linksys] callerid=linksys type=friend user=linksys secret=linksys context=default host=dynamic If I call extension 2: -- Executing BackGround(SIP/linksys-081a5678, demo-moreinfo) in new stack -- Playing 'demo-moreinfo' (language 'en') -- Executing Goto(SIP/linksys-081a5678, s|instruct) in new stack -- Goto (default,s,6) -- Executing BackGround(SIP/linksys-081a5678, demo-instruct) in new stack -- Playing 'demo-instruct' (language 'en') -- Executing WaitExten(SIP/linksys-081a5678, ) in new stack -- Timeout on SIP/linksys-081a5678, going to 't' -- Executing Goto(SIP/linksys-081a5678, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/linksys-081a5678, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/linksys-081a5678, ) in new stack All correct. Using mysql: mysql select callerid,type,name,secret,context,host,allow,disallow from sip; +--++-+-+-+-+-+--+ | callerid | type | name| secret | context | host| allow | disallow | +--++-+-+-+-+-+--+ | Linksys | friend | linksys | linksys | default | dynamic | g729;ilbc;gsm;ulaw;alaw | all | +--++-+-+-+-+-+--+ If I call extension 2: -- SIP Seeding peer from astdb: 'linksys' at [EMAIL PROTECTED]:5060 for 3600 -- Executing BackGround(SIP/linksys-081a5678, demo-moreinfo) in new stack Oct 3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to gsm Oct 3 13:16:17 WARNING[13896]: file.c:824 ast_streamfile: Unable to open demo-moreinfo (format g729): No such file or directory Oct 3 13:16:17 WARNING[13896]: pbx.c:5798 pbx_builtin_background: ast_streamfile failed on SIP/linksys-081a5678 for demo-moreinfo -- Executing Goto(SIP/linksys-081a5678, s|instruct) in new stack -- Goto (default,s,6) -- Executing BackGround(SIP/linksys-081a5678, demo-instruct) in new stack Oct 3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to gsm Oct 3 13:16:17 WARNING[13896]: file.c:824 ast_streamfile: Unable to open demo-instruct (format g729): No such file or directory Oct 3 13:16:17 WARNING[13896]: pbx.c:5798 pbx_builtin_background: ast_streamfile failed on SIP/linksys-081a5678 for demo-instruct -- Executing WaitExten(SIP/linksys-081a5678, ) in new stack Oct 3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to g729 Oct 3 13:16:17 WARNING[13896]: channel.c:2380 set_format: Unable to find a codec translation path from ulaw to g729 -- Timeout on SIP/linksys-081a5678, going to 't' -- Executing Goto(SIP/linksys-081a5678, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/linksys-081a5678, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') I only hear demo-thanks. Any hint please? Thanks for your time -- Arkaitz -- Arkaitz ___ --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] Sip configuration using mysql
Hi, I'm trying to use mysql for sip users management and i'm a bit stuck with a problem. I use asterisk-1.2.12.1 and res_config_mysql from asterisk-addons-1.2.4. The fact is that i've put a row in the mysql sip table for my linksys phone and i can make calls and receive calls with it, but it doesn't appear in sip show peers, and asterisk is unable to find files when I use that phone configured from mysql. Sep 20 15:11:24 WARNING[8347]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to slin Sep 20 15:11:24 WARNING[8347]: app_festival.c:187 send_waveform_to_channel: Unable to set write format to signed linear Sep 20 15:22:27 WARNING[8370]: channel.c:2752 ast_channel_make_compatible: No path to translate from SIP/saladino-081aa1c8(4) to SIP/linksys-a6f017a0(256) Sep 20 15:22:48 WARNING[8376]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to gsm Sep 20 15:22:48 WARNING[8376]: file.c:824 ast_streamfile: Unable to open vm-intro (format g729): No such file or directory When i configure it from sip.conf file it works perfect (i comment the entry when i want to use the mysql conf). [linksys] callerid=linksys type=friend user=linksys secret=linksys context=saladino host=dynamic The mysql part: mysql desc sip; ++--+--+-+-++ | Field | Type | Null | Key | Default | Extra | ++--+--+-+-++ | id | int(11) | NO | PRI | NULL | auto_increment | | name | varchar(80) | NO | UNI | || | accountcode| varchar(20) | YES | | NULL || | amaflags | varchar(13) | YES | | NULL || | callgroup | varchar(10) | YES | | NULL || | callerid | varchar(80) | YES | | NULL || | canreinvite| char(3) | YES | | yes || | context| varchar(80) | YES | | NULL || | defaultip | varchar(15) | YES | | NULL || | dtmfmode | varchar(7) | YES | | NULL || | fromuser | varchar(80) | YES | | NULL || | fromdomain | varchar(80) | YES | | NULL || | fullcontact| varchar(80) | YES | | NULL || | host | varchar(31) | NO | | || | insecure | varchar(4) | YES | | NULL || | language | char(2) | YES | | NULL || | mailbox| varchar(50) | YES | | NULL || | md5secret | varchar(80) | YES | | NULL || | nat| varchar(5) | NO | | no || | deny | varchar(95) | YES | | NULL || | permit | varchar(95) | YES | | NULL || | mask | varchar(95) | YES | | NULL || | pickupgroup| varchar(10) | YES | | NULL || | port | varchar(5) | NO | | || | qualify| char(3) | YES | | NULL || | restrictcid| char(1) | YES | | NULL || | rtptimeout | char(3) | YES | | NULL || | rtpholdtimeout | char(3) | YES | | NULL || | secret | varchar(80) | YES | | NULL || | type | varchar(6) | NO | | friend || | username | varchar(80) | NO | | || | disallow | varchar(100) | YES | | all || | allow | varchar(100) | YES | | g729;ilbc;gsm;ulaw;alaw || | musiconhold| varchar(100) | YES | | NULL || | regseconds | int(11) | NO | | 0 || | ipaddr | varchar(15) | NO | | || | regexten | varchar(80) | NO | | || | cancallforward | char(3) | YES | | yes || | setvar | varchar(100) | NO | | || ++--+--+-+-++ Phone row: id=2 name=linksys canreinvite=yes context=saladino dtmfmode=rfc2833 host=dynamic nat=yes secret=linksys type=peer username=linksys disallow=all allow=g729;ilbc;gsm;ulaw;alaw Other fields are NULL. Any hint? Thanks for your time. -- Arkaitz ___ --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] Sip configuration using mysql
Hi, Thanks, now i see the phone in show sip peers, I've been reading about rtcachefriends and now i understand what was the problem. But the other problem is still here :(. It seems that asterisk is unable to find any file in the system, not gsm file nor codec... nothing. It's strange since i provide the same options in sip.conf than in mysql row, but still it fails. i don't understand why. Thanks for your time On 9/20/06, Michel Vaillancourt [EMAIL PROTECTED] wrote: Arkaitz wrote: Hi, I'm trying to use mysql for sip users management and i'm a bit stuck with a problem. I use asterisk-1.2.12.1 and res_config_mysql from asterisk-addons-1.2.4. The fact is that i've put a row in the mysql sip table for my linksys phone and i can make calls and receive calls with it, but it doesn't appear in sip show peers, and asterisk is unable to find files when I use that phone configured from mysql. Try: /etc/asterisk/sip.conf [general] rtcachefriends=yes -- --Michel Vaillancourt Senior Telephony Engineer Neoxo Inc (www.neoxo.com) +1 514 395 1106 ext 117 ___ --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 -- Arkaitz ___ --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] Sip configuration using mysql
Hi, On 9/20/06, Michel Vaillancourt [EMAIL PROTECTED] wrote: Arkaitz wrote: Hi, Thanks, now i see the phone in show sip peers, I've been reading about rtcachefriends and now i understand what was the problem. But the other problem is still here :(. It seems that asterisk is unable to find any file in the system, not gsm file nor codec... nothing. It's strange since i provide the same options in sip.conf than in mysql row, but still it fails. i don't understand why. Thanks for your time Suggest you check file permissions vs the user that Asterisk is running as. Ok, I'll check tomorrow(i'm not at work now), but if the problem is the permissions i think it should fail too using sip.conf instead of mysql, i supose that the way it manages users is not related to the user that Asterisk is running as nor to the permissions of the filesystem. i am confused? Thanks for your time -- Arkaitz ___ --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
