[Asterisk-Users] A CDR issue of agent.conf createlink feature
Hi,
Asterisk version : 1.2.7.1 stable version
We try agent.conf setting of
createlink=yes
We always can not see this link value to be filled in MySQL's
table filed : userfield
But we can see the record file has been created correctly.
In debug mode, no userfiled shown in SQL command,
May 17 18:10:51 DEBUG[2889] pbx.c: Function result is
'"unknown" 2001'May 17 18:10:51 DEBUG[2889] pbx.c: Function result
is '2001'May 17 18:10:51 DEBUG[2889] pbx.c: Function result is '1'May 17
18:10:51 DEBUG[2889] pbx.c: Function result is 'sip_ps'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is 'SIP/2001-783e'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is 'Agent/1000'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is 'Queue'May 17 18:10:51 DEBUG[2889]
pbx.c: Function result is '1180'May 17 18:10:51 DEBUG[2889] pbx.c:
Function result is '2006-05-17 18:10:40'May 17 18:10:51 DEBUG[2889] pbx.c:
Function result is '2006-05-17 18:10:41'May 17 18:10:51 DEBUG[2889] pbx.c:
Function result is '2006-05-17 18:10:51'May 17 18:10:51 DEBUG[2889] pbx.c:
Function result is '11'May 17 18:10:51 DEBUG[2889] pbx.c: Function result is
'10'May 17 18:10:51 DEBUG[2889] pbx.c: Function result is 'ANSWERED'May
17 18:10:51 DEBUG[2889] pbx.c: Function result is 'DOCUMENTATION'May 17
18:10:51 DEBUG[2889] pbx.c: Function result is '2001'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is '1147860640.0'May 17 18:10:51
DEBUG[2889] pbx.c: Function result is 'agent-1000-1147860640-3.wav'May 17
18:10:51 DEBUG[2889] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.May 17 18:10:51 DEBUG[2889] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2006-05-17 18:10:40','\"unknown\" 2001','2001','1','sip_ps',
'SIP/2001-783e','Agent/1000','Queue','1180',11,10,'ANSWERED',3,'2001')May
17 18:10:51 DEBUG[2889] chan_sip.c: update_call_counter(2001) - decrement call
limit counter
Do I miss any important flag in config to enable this
field?
best regard
kaiser
___
--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 on HP DL380 G4 - chan_zap.so problems
Hi there! I compiled asterisk on a HP DL380 G4 with Suse Linux Enterprise Server 9 (gcc 3.3.3). It compiled without any errors. I also had no problems with installing my digium hardware (WC TE110P). But when I try to start asterisk, I get the following error messages: The error messages Apr 12 10:22:37 WARNING[8756]: chan_iax2.c:4796 timing_read: Unable to acknowledge zap timer .. == Parsing '/etc/asterisk/zapata.conf': Found Apr 12 10:22:37 WARNING[8756]: chan_zap.c:924 zt_open: Unable to specify channel 1: Inappropriate ioctl for device Apr 12 10:22:37 ERROR[8756]: chan_zap.c:6460 mkintf: Unable to open channel 1: Inappropriate ioctl for device here = 0, tmp-channel = 1, channel = 1 Apr 12 10:22:37 ERROR[8756]: chan_zap.c:10247 setup_zap: Unable to register channel '1-15' Apr 12 10:22:37 WARNING[8756]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Apr 12 10:22:37 WARNING[8756]: loader.c:440 load_modules: Loading module chan_zap.so failed! Can anybody help me on this? I would really appreciate that :) Luke PS: Sorry for my English Loaded modules: Module Size Used by snd_pcm_oss 65704 0 snd_pcm 112900 1 snd_pcm_oss snd_page_alloc 16264 1 snd_pcm snd_timer 32260 1 snd_pcm snd_mixer_oss 24448 1 snd_pcm_oss snd 71012 4 snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss soundcore 13536 1 snd edd 13720 0 joydev 14528 0 sg 41760 0 st 45212 0 sr_mod 21028 0 ide_cd 42628 0 cdrom 43036 2 sr_mod,ide_cd nvram 13448 0 wcte11xp28448 0 zaptel 188420 1 wcte11xp hw_random 9620 0 ehci_hcd33668 0 uhci_hcd35728 0 thermal 16648 0 processor 21568 1 thermal fan 8196 0 button 10384 0 evdev 13952 0 battery 12804 0 ipv6275580 17 ac 8964 0 raw 44064 0 usbcore 116700 4 ehci_hcd,uhci_hcd tg3 80516 0 isdn145612 0 slhc11392 1 isdn subfs 12160 1 dm_mod 59904 0 ext3123688 1 jbd 75172 1 ext3 cciss 47332 3 sd_mod 25088 0 scsi_mod120132 5 sg,st,sr_mod,cciss,sd_mod ztcfg -vv: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. My /etc/zaptel.conf loadzone=nl defaultzone=nl span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 My /etc/asterisk/Zapata.conf [channels] language=de context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no
[Asterisk-Users] Canreinvite issue
Hi , all: Anyone try sip channel with canreinvite=yes? sometimes we see a new INVITE will be send to UA immediately after user hangup the call. It makes the phone ring again after hangup. Anyone know what happen? It not always, maybe 2-5% only. But it make user crazy. Thanks... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Absolute Timeout
Hi, I dial a number with following setting: exten = _X.,1,Absolutetimeout(20)exten = _X.,2,dial(SIP/[EMAIL PROTECTED]|L(30))exten = T,1,BackGround(tt-weasels)exten = T,2,Hangup() I find Absolute time out is not working , is it normal? kaiser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Channels
In a old mailing list, someone got the trouble, anyone has idea? I am getting this when I do a: show sip channels 209.82.xxx.xxx 0071495217 2591218534@ 00103/1 unknow(d) 209.82.xxx.xxx 0041590104 0690231739@ 00103/1 unknow(d) 209.82.xxx.xxx 0070259259 3265102826@ 00103/1 unknow(d) 209.82.xxx.xxx 0071948143 1927207026@ 00103/1 unknow(d) 209.82.xxx.xxx 0022576786 1752809624@ 00103/1 unknow(d) 209.82.xxx.xxx 0070153955 0085223171@ 00103/1 unknow(d) I have about 60 of them and growing. I have submitted a ticket with my provider to let them know of this problem but I would like to clear them out w/o restarting the asterisk binary. thanks gupiter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] API manager - Redirect with ExtraChannel
Hi, We try to do something likesomone did in redirectAPI, but not fully success... This is what we did, Both channel has been setup and talking... Action: RedirectChannel: SIP/210.201.75.100-081b9170ExtraChannel: SIP/route886x-79cbExten:18Context:sipPriority:1 I have two issue: 1. Channel and Extrachannel could be the same tech channel, sip? 2. Always one certain party connected, one disconnect //Zombie ?? why?? Event: LinkChannel1: SIP/210.201.75.100-08168dd0Channel2: SIP/route886x-5550Uniqueid1: 1108739916.2Uniqueid2: 1108739925.3 Action: RedirectChannel: SIP/210.201.75.100-08168dd0ExtraChannel: SIP/route886x-5550Exten: 18Context: sipPriority: 1 Event: NewchannelChannel: AsyncGoto/SIP/route886x-5550State: UpCallerid: unknownUniqueid: 1108739972.4 Event: RenameOldname: SIP/route886x-5550Newname: SIP/route886x-5550MASQUniqueid: 1108739925.3 Event: RenameOldname: AsyncGoto/SIP/route886x-5550Newname: SIP/route886x-5550Uniqueid: 1108739972.4 Event: RenameOldname: SIP/route886x-5550MASQNewname: AsyncGoto/SIP/route886x-5550ZOMBIEUniqueid: 1108739925.3 Event: NewextenChannel: SIP/route886x-5550Context: sipExtension: 18Priority: 1Application: AnswerAppData:Uniqueid: 1108739972.4 Event: NewextenChannel: SIP/route886x-5550Context: sipExtension: 18Priority: 2Application: WaitAppData: 1Uniqueid: 1108739972.4 Response: SuccessMessage: Dual Redirect successful Event: UnlinkChannel1: SIP/210.201.75.100-08168dd0Channel2: AsyncGoto/SIP/route886x-5550ZOMBIEUniqueid1: 1108739916.2Uniqueid2: 1108739925.3 Event: HangupChannel: AsyncGoto/SIP/route886x-5550ZOMBIEUniqueid: 1108739925.3Cause: 16 We use this in the astGUIclient to transfer an active conversation(bothparties) to a meetme room:Action: RedirectChannel: Zap/73-1ExtraChannel: SIP/199testphone-1f3cExten: 8600029Context: defaultPriority: 1where 8600029 is a meetme room.Works very well.Sadly like most obscure features in Asterisk it is not documented anywherevery well. But ExtraChannel in Redirect is the only way to send both partieson a 2-party call into a meetme room so that they can be joined by a 3rdparty(without having a multi-line phone that is).MATT--- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
