Re: [Asterisk-Users] Obtaining billsecs in the dialplan after a call?
Hi,
You can use the ANSWEREDTIME variable :
exten = *244*,n,Noop(after dial duration is ${ANSWEREDTIME})
Regards,
Olivier
-
http://www.olivier-perrin.net
Le vendredi 10 février 2006 à 12:19 +0200, [EMAIL PROTECTED] a écrit :
Hi,
I'm stuck on a silly thing. I need to get the billsec CDR value after a
call. But I'm finding its always 0.
Here's my test code:
exten = *244*,1,Dial(Local/[EMAIL PROTECTED]/n,,g)
exten = *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is
${CDR(billsec)})
exten = *244*,n,Hangup
[custom-tests]
exten = test,1,Answer
exten = test,n,Playback(tt-somethingwrong)
exten = test,n,Hangup
The actual CDR record that gets posted in Master.csv looks like so:
,200,*244*,default,Exten 200 200,SIP/200-94dd,Local/[EMAIL
PROTECTED],1,Hangup,,2006-02-10
11:57:42,2006-02-10 11:57:42,2006-02-10
11:57:45,3,3,ANSWERED,DOCUMENTATION
So the duration is there just fine. But ${CDR(billsec)} remains stubbonly
0.
Now I don't really understand the CDR code 100% - but it looks like
billsec is only worked out then the cdr is posted. But there is no way to
force the cdr to be posted from the dialplan, is there?
Thanks,
Steve
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Re: [Asterisk-Users] Re: asterisk logger - urgent!!!
Hi, You just have to remove cdr_csv.so module : unload cdr_csv.so under the CLI or add noload = cdr_csv.so in /etc/asterisk/modules.conf and reload asterisk -- http://www.olivier-perrin.net Le jeudi 09 février 2006 à 11:26 -0200, Dov Bigio a écrit : I found the problem. Master.csv reached 2.0GB and since the moment this happened Asterisk went crazy! Since I am using cdr-mysql, how do I disable the use of csvs? Thank you Dov - Original Message - From: Dov Bigio To: asterisk-users@lists.digium.com Sent: Thursday, February 09, 2006 10:56 AM Subject: asterisk logger - urgent!!! Hi, Since yesterday my Asterisk 1.2.3 is displaying the following message every few seconds Asterisk Event Logger restarted Rotated Logs Per SIGXFSZ (Exceeded file size limit) This causes my log files (verbose, queue_log) to become huge with lots of logger rotate messages, but I don't know which files is exceeding size limit, since even if I delete all log files I still get this message. Any way, I have plenty of disk space and couldn't find the reason for this message. Please help me identify the issue! Dov ___ --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] 'h' in CDR
hi, 'h' is the Hangup extension. You'll find more information about this on the wiki : http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension Regards, http://www.olivier-perrin.net Le vendredi 20 janvier 2006 à 10:55 +0100, Mimmus a écrit : Hi, I'm seeing a lot of 'h' as destination numbers in my CDR logs. Some time ago I solved this problem but now I'm not able to remember anymore. Something related to match-all extension? Any help? Thanks Mimmus ___ --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] TE410P E1 Red Alarm
Hi, You could only take timing from one E1 per card. So you should use : span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 instead of : span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,1,0,ccs,hdb3,crc4 span=4,1,0,ccs,hdb3,crc4 Le dimanche 25 décembre 2005 à 03:07 -0600, Diyanat Ali a écrit : Hello! I have a TE410P quad span card with 4 E1, i am using asterisk 1.2.1, i was using it without any issues earlier with just 1 E1 on span 1 and i recently plugged in 3 more E1's, only span 1 is working, the e1 for span 1 is from a different provider then the rest, the settings are same for both, but i constantly get red alaram on the span 2,3,4, i tried all settings, including the timming source , framing, coding , signalling type etc, without any sucesss what maybe the cause of the red alarm Regards Diyanat lspci -vvv 06:01.0 Communication controller: Unknown device d161:0410 (rev 02) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR+ FastB2B- Status: Cap- 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 64 Interrupt: pin A routed to IRQ 74 Region 0: Memory at fdff (32-bit, non-prefetchable) [size=128] lsmod Module Size Used byNot tainted wct4xxp78432 124 zaptel183776 250 [wct4xxp] cat /proc/zaptel/* Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 1 TE4/0/1/1 Clear (In use) upto 16 TE4/0/1/16 HDLCFCS (In use) upto 31 TE4/0/1/31 Clear (In use) Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED 32 TE4/0/2/1 Clear (In use) upto 47 TE4/0/2/16 HDLCFCS (In use) upto 62 TE4/0/2/31 Clear (In use) Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED 63 TE4/0/3/1 Clear (In use) upto 78 TE4/0/3/16 HDLCFCS (In use) upto 93 TE4/0/3/31 Clear (In use) Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 RED 94 TE4/0/4/1 Clear (In use) upto 109 TE4/0/4/16 HDLCFCS (In use) upto 124 TE4/0/4/31 Clear (In use) ztcfg -v SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 124 channels configured. zttool Alarms Span OK T4XXP (PCI) Card 0 Span 1 RED T4XXP (PCI) Card 0 Span 2 RED T4XXP (PCI) Card 0 Span 3 RED T4XXP (PCI) Card 0 Span 4 CLI zap show status Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2RED0 0 0 T4XXP (PCI) Card 0 Span 3RED0 0 0 T4XXP (PCI) Card 0 Span 4RED0 0 0 alpha*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 alpha*CLI pri show span 2 Primary D-channel: 47 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 its the same as span for the rest as upto span 4 CLIpri intense debug span 2 Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data Sending Set Asynchronous Balanced Mode Extended same for the rest zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan = 1-15 dchan = 16 bchan = 17-31 span=2,1,0,ccs,hdb3,crc4 bchan = 32-46 dchan = 47 bchan = 48-62 span=3,1,0,ccs,hdb3,crc4 bchan = 63-77 dchan = 78 bchan = 79-93 span=4,1,0,ccs,hdb3,crc4 bchan = 94-108 dchan = 109 bchan = 110-124 loadzone=se defaultzone=se zapata.conf [channels] language=us context=sip switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes cidsignalling=dtmf cidstart=ring rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived group=1 channel = 1-15 channel = 17-31 group=2 channel = 32-46 channel = 48-62 group=3 channel = 63-77 channel =
Re: [Asterisk-Users] Error compiling asterisk addons version 1.2.0-beta2
You need to install the libmysqlclient-dev package. Search the rpm/deb/src available for your distrib and install it. Regards, Le mar 08/11/2005 à 10:44, Mohamed A. Gombolaty a écrit : Dear All, I am facing a problem in compiling the add-ons for the mysql, though the files are downloaded correctly and checked and I tried different mirrors even the cvs but yet I get those errors : app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directory cdr_addon_mysql.c:39:20: errmsg.h: No such file or directory res_config_mysql.c:51:19: mysql.h : No such file or directory res_config_mysql.c:52:27: mysql_version.h: No such file or directory res_config_mysql.c:53:20: errmsg.h: No such file or directory anyone has a clue, I used to compile it without problems Thx MAG ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] ericsson pabx and digium card TE110P
According your conf, you are in France, so i answer in french :-) Si le schéma est bien le suivant : E1(Frannce-telecom) -- PABX --- Asterisk. Et que les appels entrants sont transmis à * avec seulement 4 digits, c'est plus un problème d'opérateur que de PABX. En effet, traditionnellement France-Telecom n'envoie sur une E1 louée aux entreprises que les 4 derniers digits des appels entrants. Ce qui en général permet de savoir vers quel poste envoyer l'appel entrant. Cordialement, Le mar 08/11/2005 à 06:33, Chee Foong a écrit : Did you verify with the pbx engineer on how many digits the pbx are sending? Usually this should be the setting in the pbx. CCF -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of vador loupe Sent: Sunday, October 30, 2005 10:23 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] ericsson pabx and digium card TE110P Hi; Could some one help me: I have a problème to make call from my pabx ericsson i receive juste 4 digit from ericsson to my asterisk any idea??? thanks zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=fr defaultzone=fr zapata.conf: [channels] language=fr switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown hidecallerid=no threewaycalling=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no context=entrant group = 0 signalling=pri_net channel = 1-15 channel = 17-31 __ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] meetme conference getting error using codec g729
Hi, Seem to be a G729 licences issue. Have you buy G729 licences ? Regards, Le lun 07/11/2005 à 11:42, nr k a écrit : Hi all when i try to the conference the number i am getting the following error in asterisk console. i am using the g729 codec in asterisk and my sip devices but i can able make the call between the device. error: Nov 7 16:07:49 NOTICE[3190]: channel.c:1703 ast_set_write_format: Unable to find a path from gsm to g729 Nov 7 16:07:49 WARNING[3190]: file.c:787 ast_streamfile: Unable to open conf-onlyperson (format g729): No such file or directory regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] PSTN CALLER ID FRANCE TELECOM
Hi Guillaume, You can find a solution here : http://www.voip-info.org/wiki/view/CID+Issues+with+some+Siemens+DECT+phones+in+France Was working for me few month ago. Regards Olivier Le lun 10/10/2005 à 19:28, guillaume a écrit : Hi all I try to get the caller id of a incoming call through a X100P generic card. I have tried many configuration on the zapata.conf, but i never succeed to have a correct CALLERIDNUM. What is the cid signaling provided by FranceTelecom (v23 ?) Is there some specific stuff to do ? Could you help me please ? Guillaume The provider is france telecom, The card is a X100P asterisk -V = 1.0.9 The error message is : Oct 10 20:17:02 WARNING[702]: chan_zap.c:5476 ss_thread: Calleerror on channel 'Zap/1-1' my zapata.conf is ~ context=from-ft language=fr signalling=fxs_ks busydetect=yes busycount=1 callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=6.0 txgain=4.0 immediate=no callerid=asreceived musiconhold=default channel = 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] AGI problem with library path
You can add your library path in the /etc/init.d/asterisk script and restart asterisk via service. or You also can 1/ add your library path in your /etc/ld.so.conf 2/ type ldconfig as root. 3/ restart asterisk via service Hi List, My AGI seems work well in asterisk -vvvc mode, other than that it doesn't work. Its seems to me, when I run asterisk as daemon (service asterisk start .. on fc4), it doesn't know about my library path. How can pass libray path to my AGI script or asterisk? Thanks___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Digium Generic Boards - Low Prices / High Quality.
Please two emails in two days, it's enought. Don't send an other one tomorrow or use -biz list Le mer 10/11/2004 à 00:03, Richard Moore a écrit : Hello list , As announced before we're starting selling only E100P based boards. T100P boards and other digium-like products will be available in 20-30 days. All products are being tested for our engineers. We're developing our website with all products and we'll include a web shopping for global sales. At this momment , i'm at Brazil negotiating a distribution channel for latin america. 500 units were sold only today. (E100P) So, please wait for our products to be released.! don't waste money :) Best Regards, Richard _ MSN Hotmail, o maior webmail do Brasil. http://www.hotmail.com ___ 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] New Release Asterisk-Stat V 1.3
Great job ! I use it everyday to see CDR of 3 Asterisk servers. Thx a lot Le mar 09/11/2004 à 16:01, Areski a écrit : New Release Asterisk-Stat ! http://www.areski.net/asterisk-stat-v1_3 Asterisk-Stat is a Web CDR viewer. To make easy the way to consult all your CDR and also to compare the traffic during different days. For an easiest analyse of your CDR ! - Support mysql postgresql - need GD library and jpgraph_lib (included here) NEWS : - Export CDR to PDF - Export CDR to CSV - New criteria based on the duration - Graph correction (Thanks to Jerome :) - fews other bugs fixed... More details - http://www.areski.net/asterisk-stat-v1_3 Kinds regards, /Areski ___ 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] Help configuring an Wildcard E100P
Hi,
try this :
Dial(Zap/g1/0553024039)
instead of
Dial(Zap/g1,0553024039)
Le jeu 18/03/2004 à 16:12, Alessio Focardi a écrit :
Hi !
I need a quick help configuring an Wildcard E100P ...
Inbound calls are working ok, but I can not call out, dialing 20 only
gets me a line dial tone, but no call is made; same stuff with _0.
direct dialing
Please provide some suggestions if you have ! TNX !
This is my actual config
zapata.conf
[channels]
signalling=pri_cpe
switchtype=euroisdn
group=1
context=default
channel = 1-15
channel = 17-31
zaptel.conf
loadzone = us
defaultzone = us
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
extentions.conf
[general]
static=yes
writeprotect=no
[globals]
[default]
exten = 20,1,Answer ; Answer the line
exten = 20,2,Dial(Zap/g1,0553024039)
exten = _0.,1,Dial,Zap/g1,${EXTEN:1}|45|r
exten = _0.,2,Congestion
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[Asterisk-Users] Playtones and ISDN question
Hi everybody, Is it possible to use Playtones without Answer a call ? It's for a callback application. I want to play a tone to inform the user if Asterisk callback his number and an other if his calerid is refused. It works with iax2 and not with Euro-Isdn (E100P) -- [indication.conf] [general] country=fr [fr] description = France ringcadance = 1500,3500 ; Dialtone can also be 440+330 dial = 440 busy = 440/500,0/500 ring = 440/1500,0/3500 ; XXX I'm making up the congestion tone XXX congestion = 440/250,0/250 ; XXX I'm making up the call wait tone too XXX callwait = 440/300,0/1 ; XXX I'm making up dial recall XXX dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 ; XXX I'm making up the record tone XXX record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330 --- in /etc/zaptel.conf : loadzone=fr defaultzone=fr ___ 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] E100P : Pb with outgoing calls
I use a E100P in France with a french operator E1. I can receive calls via the E1 and tranfer them to a VoIP phone, play IVR etc But outgoing calls doesn't work at all. I receive a RELEASE COMPLETE just after the SETUP. There is no pb with the operator (the E1 work well with an other Pbx). Here a call trace. Anyone have an idea ? (g1 is my group name for the 30 channels) -- Accepting call from '147241527' to '5797' on channel 21, span 1 -- Executing Dial(Zap/21-1, Zap/g1/3361100) in new stack -- Making new call for cr 32784 Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 2 (reference 16/0x10) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '147241527' ] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3361100' ] Sending Complete (len= 0) -- Called g1/3361100 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32784/0x8010) (Terminator) Message type: RELEASE COMPLETE (90) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: International network (7) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (6) ] -- Processing IE 8 (Cause) -- Channel 1, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' == No one is available to answer at this time ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
