[asterisk-users] memory leak on asterisk 1.6.0.6
for the second time i'm asking in this forum, somebody help me my asterisk box have a problem with memory leak. I'm scheduling to rstart the box to fix this problem but any cleverer suggest to fix this? coz this issue causing another problem to my AGI application... thankyou before _ NEW! Get Windows Live FREE. http://www.get.live.com/wl/all___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] integrating CTI
Hi there, I am integrating CTI functionality to my java application and using Asterisk as PBX. I need some advices as to whether I am in the right track. • Asterisk server is configured and working fine. • I have a generic sip hard phone • I have X-lite soft phone installed. To provide for CTI functionality, I am utilizing the asterisk manager API, I designed a java application that can do the following: • Agent login • Originate call • Hang up • Transfer call • Record on server Is this the right approach, or I should do it differently, please advice? thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] memory leak on asterisk 1.6.0.6
If you are not a developer and are not capable of identifying the leak - broadly or specifically, at the very least you need to provide details, log output, and/or other material evidence relevant to the circumstances of the memory leak. There is absolutely nothing productive anyone can tell you in response to, I've got a memory leak, and nothing more. frangky robert wrote: for the second time i'm asking in this forum, somebody help me my asterisk box have a problem with memory leak. I'm scheduling to rstart the box to fix this problem but any cleverer suggest to fix this? coz this issue causing another problem to my AGI application... thankyou before Make the most of what you can do on your PC and the Web, just the way you want. Windows Live http://www.get.live.com/wl/all ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk automatically closing the file descriptor
I am using Asterisk manager API and have login-ed (connected) to it through 2 FDs: 1 for making calls and 2nd for receiving responses to calls like call hangup, fail success etc. After making some calls, asterisk is closing the FD through which I send the calls, any idea on why this is happening ? -- http://uttre.wordpress.com/2008/05/14/the-lost-love-of-mine/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Memory leak on asterisk 1.6.0.6
On Fri, May 22, 2009 at 08:04:29AM +, frangky robert wrote: Hi everyones, I have a production server using asterisk 1.6.0.6 using php as an IVR and mssql server (on other machine) My server attached a Sangoma A104 card (4xT1 card) i have a problem with memory leak on that server and causing a delay on IVR prompt. Could you please be more specific? Can you provide a trace from the logs (they have timestamps)? (Thats my assumption, memory leak problem) What makes you think that this is a memory leak? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
- Original Message - From: Jason Aarons (US) To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, May 22, 2009 10:32 PM Subject: Re: [asterisk-users] DTMF Then if it's a IP interface (SIP, etc) have you tried a sniffer trace (wireshark, etc) to verify the packets are being sent correctly to carrier? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, May 22, 2009 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF Is this inbound calls to your automated attendant? Or Outbound calls to say a bank ivr out in the pstn? What direction? What is your interface/carrier? T1, SIP, H32? And what method are you using for DTMF? Eg inband, out of band, what rfc, etc? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Friday, May 22, 2009 3:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF We are facing alot of problem in the DTMF. At times we are unable to do the verification because whenever we press the numbers for verification it does not detects and at times it detects the wrong number for instance if the customer is having the phone no. as 1234567890 it will detect 123467890 or 234567890 . And we also face the problem that the line get disconnected while doing the verification and also at times the conference is not working . -- Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open source SIP client
Pascal Bruno wrote: It seems like a few people including me DID understand what Dhaval meant, or maybe some people used they common sense and their intelligence to understand what somebody who's english is not the primary language wanted to say and put some effort to guide or help someone in the community getting to the right direction instead of trying to put him down. I think a few others need to consider investigating more deeply the basic mechanics of understanding written English, or should themselves research what some collections of syllables intend to convey. I also think if they were that good, why not provided some english tutoring instead of putting people down. Good luck in you research Dhaval! Well said. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] /etc/asterisk/startup.d
Tzafrir Cohen schrieb: On Fri, May 22, 2009 at 01:33:59PM +0200, Philipp Kempgen wrote: Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? What would you put there? Scripts to generate Asterisk config files in /etc/asterisk I guess. Or scripts to log warnings to syslog if the configuration is insecure (MySQL does that on Debian). Or maybe scripts to open some ports on a firewall. Well, no, there should be stop scripts as well then, so forget about the firewall. OTOH: The scripts could be called with an argument just like init scripts (start|stop|restart|...). I'm not quite sure if that would be a useful thing to have or if such tasks should rather be done by interdependent init scripts, i.e. Required-Start, Required-Stop, Should-Start, Should-Stop headers. When should it be run? Right before /etc/init.d/asterisk is about to (re?)start asterisk. As which user? Good question. Obviously either as root because /etc/init.d/asterisk is run by root or as Asterisk's runuser which is likely to be one of root or asterisk. root would buy us more flexibility :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Hi everyone, I just found this thread, which is amazing as I'm on my first go with asterisk and so far I've been pulling my hair out for the last week :-) I have 2 questions which were raised while this fault was being debugged. 1) Gordon says:- Is this a place where you get a polarity reversal event on call startup? In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing Well I've got an NTL phone line, can anyone tell me what to use for that? 2) Do I still need the same 2pin cable? Because I've been to Maplins too and bought one that I thought was right, but this one is a 4pin too. Can anyone tell me which pins on their 2pin cable are connected at each end? I'll bodge my cable until it works and then get a proper one once I'm sure. Thanks in advance. Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Load, Asterisk Disconnected
Ok
Thanks a lot for your reply.
Date: Mon, 18 May 2009 14:51:58 +0300
From: a...@iq-labs.net
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Queue Load, Asterisk Disconnected
2009/5/17 Torintino T torinti...@hotmail.com:
I have Asterisk 1.2.29, Zaptel 1.2.24 , TE 121P Digium Card, and Freepbx
Setup for a queue up to 15 agents through a PRI line, it was working fine
for more than 1 year, suddenly, when there is a load on the queue, the
asterisk service disconnects and the calls are dropped. And the service
starts again after few seconds, and so on.
I am not using fax.
I checked PRI by zttool and there are no alarms.
The cdr logs
2195,2,,Busy,,2009-05-13 10:06:53,,2009-05-13 10:06:55,2,0,NO
ANSWER,DOCUMENTATION
,0225167604,237,from-internal,0225167604
0225167604,Local/2...@from-internal-a07f,2,,Busy,,2009-05-13
10:06:55,,2009-05-13 10:06:57,2,0,NO ANSWER,DOCUMENTATION
,0225167604,229,from-internal,0225167604
0225167604,Local/2...@from-internal-c662,2,,Busy,,2009-05-13
10:06:57,,2009-05-13 10:06:59,2,0,NO ANSWER,DOCUMENTATION
,0225167604,224,from-internal,0225167604
0225167604,Local/2...@from-internal-3a3a,2,,Busy,,2009-05-13
10:06:59,,2009-05-13 10:07:00,1,0,NO ANSWER,DOCUMENTATION
/usr/sbin/safe_asterisk: line 57: 14229 Segmentation fault (core dumped)
${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
cat: /var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.
Verbosity logs:
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090513-092729|1242196049.184: Inbound recording enabled.
recordingcheck|20090513-092729|1242196049.184: CALLFILENAME=1242196049.184
-- AGI Script recordingcheck completed, returning 0
-- Executing Monitor(Local/2...@from-internal-b759,2,
wav49|1242196049.184| mb) in new stack
-- Executing Macro(Local/2...@from-internal-b759,2, dial|30|Ttr|211) in
new stack
-- Executing AGI(Local/2...@from-internal-b759,2, dialparties.agi) in
new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
-- dialparties.agi: priority is 1
dialparties.agi: Caller ID name is '0227559600' number is '0227559600'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 211 to extension map
-- dialparties.agi: Extension 211 cf is disabled
dialparties.agi: Extension 211 has do not disturb enabled
-- AGI Script dialparties.agi completed, returning 0
-- Executing NoOp(Local/2...@from-internal-b759,2, Returned from
dialparties with no extensions to call) in new stack
-- Executing Set(Local/2...@from-internal-b759,2, DIALSTATUS=BUSY) in
new
stack
-- Executing GotoIf(Local/2...@from-internal-b759,2, 1?s-BUSY|1) in new
stack
-- Goto (macro-exten-vm,s-BUSY,1)
-- Executing NoOp(Local/2...@from-internal-b759,2, Extension is reporting
BUSY and has no Voicemail) in new stack
-- Executing Busy(Local/2...@from-internal-b759,2, ) in new stack
-- Local/2...@from-internal-b759,1 is busy
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2'
== Spawn extension (ext-queues, 100, 7) exited non-zero on 'Zap/25-1'
-- Hungup 'Zap/25-1'
-- Stopped music on hold on Zap/27-1
-- Playing periodic announcement
-- Playing 'custom/Busy' (language 'en')
-- Called Local/2...@from-internal/n
-- Executing Macro(Local/2...@from-internal-e5d7,2, exten-vm|novm|221)
in
new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2, user-callerid) in
new
stack
-- Executing Set(Local/2...@from-internal-e5d7,2, AMPUSER=) in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 0?109) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2, EMERGENCYCID=) in new
stack
-- Executing Set(Local/2...@from-internal-e5d7,2, AMPUSERCIDNAME=) in
new
stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 1?7) in new stack
-- Goto (macro-user-callerid,s,7)
-- Executing NoOp(Local/2...@from-internal-e5d7,2, Using CallerID
0227559600 0227559600) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2, FROMCONTEXT=exten-vm)
in new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2,
record-enable|221|IN)
in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 0
Re: [asterisk-users] PSTN Connection
Brent Vrieze wrote: Lyle Giese wrote: Manoj Panicker - FOES wrote: Hi Which is the best interface card to connect* PSTN* line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj That's a wide open question. How many lines? What kind of lines? What country are you in? What options are availible to you? I only have three incoming lines for a soho Asterisk install. I decided on a T1 card and picked up a used channel bank on ebay. Not the cheapest way, but it has served me very well. You are not going to get much help unless you define the problem better. Lyle Giese LCR Computer Services, Inc. HI, OK, I'm going to chime in on this one as I am going to set up an Asterisk system for our volunteer ambulance service. As a part of the Emergency Services we need to maintain a POTS line as redundancy and due to the fact that with an old style phone I don't need power for the phone to work. I plan on using a SIP provider for the rest of our phone needs. If not for the emergency services part I would go completely SIP based. Anyway I would need a FXO/FXS card for use in the US. Only one line so I don't need any of the fancy 4 line systems. I have heard you can use certain modems to do this but I would like what I am doing to be seamless and not require hacking at a problem for hours to save $50. I just want it to work quick and easy. I am unsure what you mean by What kind of lines? and What options are availible to you?. Maybe that is part of asking this question, to get some info about the phone system too. Any help would be grand. Thanks Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Brent You have defined what you are going to do, basically a small system and only need one POTS line. You could also use an ATA to convert a POTS to SIP to go into the Asterisk box. That would probably be a more supportable solution as those devices don't appear to be disappearing off the market like that modem solution is. Then if in a couple of years, lightening takes out the converter, you have a purchasable solution. You can also do this with Digium cards. Lyle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.0.9 sip.c: Serious Network Trouble ??
I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend. I'm getting: [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data . What does this mean? What do i do about it? sip worked fine in 1.4.24.1. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
On Sat, May 23, 2009 at 04:02:57PM +0100, Dunc wrote: In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing https://issues.asterisk.org/view.php?id=9096 ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.0.9: Unknown signalling method 'pri_cpe' ??
I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card. I can't make any connection over the T1. From CLI: ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling method 'pri_cpe' at line 37. cat chan_dahdi.conf cat chan_dahdi.conf [trunkgroups] [channels] language=en ;internationalprefix = 00 ;nationalprefix = 0 context=from-pstn switchtype=national pridialplan=local ;priindication=outofband usecallerid=yes callerid=asreceived hidecallerid=no callwaiting=yes ;usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group=0 callgroup=0 pickupgroup=0 ;immediate=no relaxdtmf=yes ;echotraining=yes ;echocancel=yes ;echocancelwhenbridged=no ;facilityenable=yes musiconhold=default ;overlapdial=yes ;immediate=no jbenable=yes txgain=0.0 rxgain=0.0 signalling = pri_cpe channel = 1-8 sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] /etc/asterisk/startup.d
On Sat, May 23, 2009 at 04:43:52PM +0200, Philipp Kempgen wrote: Tzafrir Cohen schrieb: On Fri, May 22, 2009 at 01:33:59PM +0200, Philipp Kempgen wrote: Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? What would you put there? Scripts to generate Asterisk config files in /etc/asterisk I guess. Or scripts to log warnings to syslog if the configuration is insecure (MySQL does that on Debian). When exactly should those be run? E.g.: asterisk -rx 'restart now' does not get them run. Do you want to guarantee some script to be run before Asterisk is started? Should it be run on a reload? On a logger-reload action? Or maybe scripts to open some ports on a firewall. Well, no, there should be stop scripts as well then, so forget about the firewall. OTOH: The scripts could be called with an argument just like init scripts (start|stop|restart|...). I'm not quite sure if that would be a useful thing to have or if such tasks should rather be done by interdependent init scripts, i.e. Required-Start, Required-Stop, Should-Start, Should-Stop headers. When should it be run? Right before /etc/init.d/asterisk is about to (re?)start asterisk. As which user? Good question. Obviously either as root because /etc/init.d/asterisk is run by root or as Asterisk's runuser which is likely to be one of root or asterisk. root would buy us more flexibility :-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.9: Unknown signalling method 'pri_cpe' ??
On Sat, May 23, 2009 at 12:23:50PM -0400, sean darcy wrote: I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card. I can't make any connection over the T1. From CLI: ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling method 'pri_cpe' at line 37. chan_dahdi was built without support for libpri . $ ldd /usr/lib/asterisk/modules/chan_dahdi.so linux-vdso.so.1 = (0x7fffc17ff000) libtonezone.so.2.0 = /usr/lib/libtonezone.so.2.0 (0x7fafb926d000) libpri.so.1.4 = /usr/lib/libpri.so.1.4 (0x7fafb9044000) libss7.so.1 = /usr/lib/libss7.so.1 (0x7fafb8e2d000) libpthread.so.0 = /lib/libpthread.so.0 (0x7fafb8c11000) libc.so.6 = /lib/libc.so.6 (0x7fafb88be000) libm.so.6 = /lib/libm.so.6 (0x7fafb863a000) /lib64/ld-linux-x86-64.so.2 (0x7fafb9762000) As you can see, in my case both PRI and SS7 signalling are expected to be supported. Another handy test: $ strings /usr/lib/asterisk/modules/chan_dahdi.so | grep pri_cpe pri_cpe Make sure you have libpri installed when Asterisk is configured. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Tzafrir Cohen wrote: On Sat, May 23, 2009 at 04:02:57PM +0100, Dunc wrote: In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing https://issues.asterisk.org/view.php?id=9096 ? Thanks for the link, at the moment my stuff doesn't work properly at all though, it's not just the caller ID stuff. I'm guessing it's down to the cable though now after reading the previous posts on this thread. If someone knows exactly which pins at each end I need to connect with a 2 wire cable that would be amazing (UK NTL phone line.) (Bueller, anyone? :-) ) Once I have incoming calls working then I think I'll be back fixing the caller ID stuff :) Cheers, Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Dunc On Sat, May 23, 2009 at 12:43 PM, Dunc d...@lemonia.org wrote: Tzafrir Cohen wrote: On Sat, May 23, 2009 at 04:02:57PM +0100, Dunc wrote: In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing https://issues.asterisk.org/view.php?id=9096 ? Thanks for the link, at the moment my stuff doesn't work properly at all though, it's not just the caller ID stuff. I'm guessing it's down to the cable though now after reading the previous posts on this thread. If someone knows exactly which pins at each end I need to connect with a 2 wire cable that would be amazing (UK NTL phone line.) (Bueller, anyone? :-) ) Once I have incoming calls working then I think I'll be back fixing the caller ID stuff :) I've some experience with openvox cards, what card are you using? What problems are you having? Cheers, -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] /etc/asterisk/startup.d
On Sat, 23 May 2009, Philipp Kempgen wrote: Tzafrir Cohen schrieb: As which user? Good question. Obviously either as root because /etc/init.d/asterisk is run by root or as Asterisk's runuser which is likely to be one of root or asterisk. root would buy us more flexibility :-) And insecurity. I'd vote for the user Asterisk will be running as. You can always use sudo in the script if needed. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Tiago Durante wrote: Dunc On Sat, May 23, 2009 at 12:43 PM, Dunc d...@lemonia.org wrote: Tzafrir Cohen wrote: On Sat, May 23, 2009 at 04:02:57PM +0100, Dunc wrote: In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing https://issues.asterisk.org/view.php?id=9096 ? Thanks for the link, at the moment my stuff doesn't work properly at all though, it's not just the caller ID stuff. I'm guessing it's down to the cable though now after reading the previous posts on this thread. If someone knows exactly which pins at each end I need to connect with a 2 wire cable that would be amazing (UK NTL phone line.) (Bueller, anyone? :-) ) Once I have incoming calls working then I think I'll be back fixing the caller ID stuff :) I've some experience with openvox cards, what card are you using? What problems are you having? Cheers, Hi Tiago, I have an OpenVox A400P11, it shows up like this... eddie ~ # lsdahdi ### Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 FXSFXSKS (EC: MG2) RED 2 FXOFXOLS (EC: MG2) Use of uninitialized value in string eq at /usr/lib64/perl5/site_perl/5.8.8/Dahdi/Chans.pm line 221. 3 unknown Use of uninitialized value in string eq at /usr/lib64/perl5/site_perl/5.8.8/Dahdi/Chans.pm line 221. 4 unknown eddie ~ # I'm pretty sure that the RED alarm is a bad thing. While googling about this error from the asterisk console *CLI [May 23 18:14:02] NOTICE[4469]: chan_dahdi.c:8164 handle_init_event: Alarm cleared on channel 1 [May 23 18:14:03] WARNING[4469]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 1: Red Alarm I discovered this thread on the mailing list, and so signed up and mailed in with the same subject. It didn't link them together though it seems, so here's a URL http://www.mail-archive.com/asterisk-users@lists.digium.com/msg223429.html If you read this specific post, and the last few before http://www.mail-archive.com/asterisk-users@lists.digium.com/msg223523.html it seems that a 2-pin cable from the wall socket to the card is required. Now, I was warned about cables and did my best to get the one that sounded right, however mine definitely has 4 pins. So my 2 questions are 1) What are the pinouts for the 2 pin cable, and I'll make my own for now (For bonus points, unless the wires are crossed over, what possible difference could it make when the TDM card only has 2 pins anyway?) 2) Is this the correct cable for NTL too? I think I should find out definite answers to the above before I worry any further about the card and Asterisk :-) Thanks for getting back to me, hope you can help. Cheers, Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Hi Dunc, On Sat, May 23, 2009 at 1:23 PM, Dunc d...@lemonia.org wrote: Tiago Durante wrote: Dunc On Sat, May 23, 2009 at 12:43 PM, Dunc d...@lemonia.org wrote: Tzafrir Cohen wrote: On Sat, May 23, 2009 at 04:02:57PM +0100, Dunc wrote: In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing https://issues.asterisk.org/view.php?id=9096 ? Thanks for the link, at the moment my stuff doesn't work properly at all though, it's not just the caller ID stuff. I'm guessing it's down to the cable though now after reading the previous posts on this thread. If someone knows exactly which pins at each end I need to connect with a 2 wire cable that would be amazing (UK NTL phone line.) (Bueller, anyone? :-) ) Once I have incoming calls working then I think I'll be back fixing the caller ID stuff :) I've some experience with openvox cards, what card are you using? What problems are you having? Cheers, Hi Tiago, I have an OpenVox A400P11, it shows up like this... eddie ~ # lsdahdi ### Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 FXS FXSKS (EC: MG2) RED 2 FXO FXOLS (EC: MG2) Use of uninitialized value in string eq at /usr/lib64/perl5/site_perl/5.8.8/Dahdi/Chans.pm line 221. 3 unknown Use of uninitialized value in string eq at /usr/lib64/perl5/site_perl/5.8.8/Dahdi/Chans.pm line 221. 4 unknown eddie ~ # I'm pretty sure that the RED alarm is a bad thing. While googling about this error from the asterisk console *CLI [May 23 18:14:02] NOTICE[4469]: chan_dahdi.c:8164 handle_init_event: Alarm cleared on channel 1 [May 23 18:14:03] WARNING[4469]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 1: Red Alarm I discovered this thread on the mailing list, and so signed up and mailed in with the same subject. It didn't link them together though it seems, so here's a URL http://www.mail-archive.com/asterisk-users@lists.digium.com/msg223429.html If you read this specific post, and the last few before http://www.mail-archive.com/asterisk-users@lists.digium.com/msg223523.html How is your configuration? Please post here. At first I thought you were using a A1200P, this card (which is very good) gave some headache some weeks ago, because it wasn't working properly with dahdi. I didn't have a chance to test it again. It's because for this card you need a module from Openvox. The A400P card that they made is a clone of the TDM and should work fine with DAHDI. For this card you shouldn't need any module from Openvox. it seems that a 2-pin cable from the wall socket to the card is required. Now, I was warned about cables and did my best to get the one that sounded right, however mine definitely has 4 pins. So my 2 questions are 1) What are the pinouts for the 2 pin cable, and I'll make my own for now (For bonus points, unless the wires are crossed over, what possible difference could it make when the TDM card only has 2 pins anyway?) 2) Is this the correct cable for NTL too? This line that you're using, can you use a regular analog phone to make call through it? I don't have any server running in UK, only USA and Latin America... But I'll assume the cabling is the same... If so you should have a connector like this: http://img.zdnet.com/techDirectory/RJ11.GIF Test your line with a regular phone. Make sure it works fine. Also make sure not to connect the PSTN line on the FXS card, you can 'burn' your FXS doing that. I think I should find out definite answers to the above before I worry any further about the card and Asterisk :-) I agree... =) Thanks for getting back to me, hope you can help. No prob, I hope too! Cheers! -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
Which one to download for CentOS ? http://www.wireshark.org/download.html#thirdparty On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote: We are facing alot of problem in the DTMF. At times we are unable to do the verification because whenever we press the numbers for verification it does not detects and at times it detects the wrong number for instance if the customer is having the phone no. as 1234567890 it will detect 123467890 or 234567890 . And we also face the problem that the line get disconnected while doing the verification and also at times the conference is not working . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.9: Unknown signalling method 'pri_cpe' ??
Tzafrir Cohen wrote: On Sat, May 23, 2009 at 12:23:50PM -0400, sean darcy wrote: I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card. I can't make any connection over the T1. From CLI: ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling method 'pri_cpe' at line 37. chan_dahdi was built without support for libpri . $ ldd /usr/lib/asterisk/modules/chan_dahdi.so linux-vdso.so.1 = (0x7fffc17ff000) libtonezone.so.2.0 = /usr/lib/libtonezone.so.2.0 (0x7fafb926d000) libpri.so.1.4 = /usr/lib/libpri.so.1.4 (0x7fafb9044000) libss7.so.1 = /usr/lib/libss7.so.1 (0x7fafb8e2d000) libpthread.so.0 = /lib/libpthread.so.0 (0x7fafb8c11000) libc.so.6 = /lib/libc.so.6 (0x7fafb88be000) libm.so.6 = /lib/libm.so.6 (0x7fafb863a000) /lib64/ld-linux-x86-64.so.2 (0x7fafb9762000) As you can see, in my case both PRI and SS7 signalling are expected to be supported. Another handy test: $ strings /usr/lib/asterisk/modules/chan_dahdi.so | grep pri_cpe pri_cpe Make sure you have libpri installed when Asterisk is configured. Got libpri-1.4.9, rebuilt asterisk, now seems to work. Thanks for the quick response. Puzzling that I built 1.4.25 earlier, and that Just Worked. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] /etc/asterisk/startup.d
On Sat, May 23, 2009 at 10:20:21AM -0700, Steve Edwards wrote: On Sat, 23 May 2009, Philipp Kempgen wrote: Tzafrir Cohen schrieb: As which user? Good question. Obviously either as root because /etc/init.d/asterisk is run by root or as Asterisk's runuser which is likely to be one of root or asterisk. root would buy us more flexibility :-) And insecurity. I'd vote for the user Asterisk will be running as. You can always use sudo in the script if needed. Depends what the script needs to do. It's the sysadmin that adds those scripts. If the script needs to check something with mysql, as in the example given by the OP, running it as asterisk is pointless. If the sysadmin wants to run something as asterisk, there's always su -c -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
On Sat, 23 May 2009, David @ULC wrote: Which one to download for CentOS ? http://www.wireshark.org/download.html#thirdparty sudo yum -y install wireshark wireshark-gnome Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] integrating CTI
On Sat, May 23, 2009 at 2:41 AM, peace keeper peacekeeperacco...@gmail.com wrote: Hi there, I am integrating CTI functionality to my java application and using Asterisk as PBX. Is this the right approach, or I should do it differently, please advice? I don't know about your application, so I have no idea whether you should do it differently. Nothing you said seems inherently wrong. Some people don't like AMI and prefer other approaches like call files, or AGI. Try putting some calls through and see what happens. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CCM, CME Integration
On Wed, May 20, 2009 at 12:44 AM, Arun Kumar arunv...@gmail.com wrote: here is my problem: when I call from 6004 to my cme extension 4615, on 4615 I've configured noans timeout to 15 and then it goes to my unity express (cue) for voicemail so when I call my cme extension it rings for few seconds and then on my asterisk cli I see 500 Internal Server Error back from my CCM IP and getting standard asterisk message saying all circuits are busy now . as per my understanding it should go to my cue. You need to enable better debugging on the Cisco side. You shouldn't be getting a 500 internal server error. You need to debug the Cisco and find out what it says besides 500 internal server error. There should be logging for an error like that. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT]I like this community
Hi @ all, i like this community, i don't think that there is any place on this planet from where emails are not coming directed to this community, if governments were profiting to each other like the members of this community do, there would be no poor on this planet, there would be no war on this planet, there would no deseases on this planet, Thanks to everybody, warm regards, 2R ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
Thanks a lot but which one to download before installing ? On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote: We are facing alot of problem in the DTMF. At times we are unable to do the verification because whenever we press the numbers for verification it does not detects and at times it detects the wrong number for instance if the customer is having the phone no. as 1234567890 it will detect 123467890 or 234567890 . And we also face the problem that the line get disconnected while doing the verification and also at times the conference is not working . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail and remote directory with SSHFS
Hello, The voicemails are sent over to an independent server to save server resources (harddisk writing, harddisk space, etc.) and allocate more bandwidth to live RTP calls. The servers are located in different locations, with each one having an independent public IP address. Accordingly, I need to mount the voicemail directory on both servers. Thanks, Elliot On Fri, May 22, 2009 at 3:32 PM, Jeff LaCoursiere j...@jeff.net wrote: Lets start from the beginning. Why are using a network share for your voicemail in the first place? j On Fri, 22 May 2009, Elliot Murdock wrote: Hello Matt, I do agree with you that NFS is that UNIX standard for network filesystems and that what should essentially be used. However, I shied away from using it, because on the surface it looks too complicated to secure properly. It uses many ports, dynamic ports, different background daemons, etc. As I stated before, to mount one or two directories, it is just not worth the trouble to set up a NFS filesystem. Accordingly, I figured I would go from bottom up, starting with sshfs, samba (which uses only 445 and 139, straightforward config file), and then if those don't work out go through the trouble of setting up a NFS filesystem. If you know of any documents that simplify the NFS (not just how to set up a simple mount, but a full tutorial that describes how it works and how to fully secure it), then I would be more than happy to implement it. Later, Elliot On 5/21/09, Matt Watson m...@mattgwatson.ca wrote: Not that I;m exactly a big fan of NFS but... why would you choose to implement a filesystem that was designed to emulate Windows shares for your UNIX-type environment? You have to kind of expect odd problems like this when you choose to use things for other than their intended purpose. Samba I would say is probably alot more focused on providing storage shares for Windows desktop clients, not for UNIX-type clients. Sure there is some support to do what you want, but just keep in mind that similiar to using sshfs like you were trying before, Samba, was really not designed to be used by UNIX clients. You've already found the most obvious reason... case sensative filenames - which Windows does not support, and UNIX programs expect filesystems on your UNIX machine *will* support it. That seems kind of like me deciding to use ntfs on a local partition on linux box instead of ext3/4, jfs, reiserfs, etc. -- Matt On Thu, May 21, 2009 at 5:06 AM, Elliot Murdock murdo...@gmail.com wrote: Hello! Thanks...I set up a Samba mount, which works ok, except that Asterisk confuses a wave file as a wav49 file. I think it may have something do with the way Samba supports case sensitivity. Since Windows is not very aggressive when it comes to being case sensitive, I am thinking that Samba is saving files with the last three characters, wav, as uppercase, WAV. What is the procedure to ensure all the files are saved as is in Samba? Thanks, Elliot On Thu, May 14, 2009 at 5:12 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Thursday 14 May 2009 08:14:17 Elliot Murdock wrote: The problem is a file locking problem that Asterisk needs to make changes to the directory. I was initially shying away from NFS and Samba, because I prefer to avoid any sort of security issues with only remotely mounting one or two directories. NFS and Samba are designed for larger applications, which makes those types of technology worthwhile. No, they're both designed as filesystems, which makes typical things like locking possible. SSH is designed as a communications medium, and someone has hacked filesystem support on top of it (poorly, apparently). SSHFS was never designed to be used in server production environments and should not be used there. I am wondering if there is any way to disable Asterisk's request to lock the directory. I know this may cause some loss in data, but for the volume voicemail receives, it should be rare enough that would make this approach an option. There is not. Use a real filesystem that supports file locking (or really, file linking, which is how the locking is implemented) procedures. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
On Sat, 23 May 2009, Dunc wrote: Hi everyone, I just found this thread, which is amazing as I'm on my first go with asterisk and so far I've been pulling my hair out for the last week :-) I have 2 questions which were raised while this fault was being debugged. 1) Gordon says:- Is this a place where you get a polarity reversal event on call startup? In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing Well I've got an NTL phone line, can anyone tell me what to use for that? Throw it away and get a BT one ;-) I'd suggest running with verbose =3 and seeing what it says. However I have 2 clients with Teleworst lines and I've never been able to make caller ID work on them, even though teleworst insist they are providing caller ID.. 2) Do I still need the same 2pin cable? Because I've been to Maplins too and bought one that I thought was right, but this one is a 4pin too. Can anyone tell me which pins on their 2pin cable are connected at each end? I'll bodge my cable until it works and then get a proper one once I'm sure. I think part of this same thread had something about modem vs. ordinary cables - however I put in a 4-pin modem cable to see what happenes and it continued to work as before, so I'm personally not convinced about that one... ie. I've not had a cable that didn't work. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
On Sun, 24 May 2009, David @ULC wrote: Thanks a lot but which one to download before installing ? Which one what? Are you referring to sudo yum -y install wireshark wireshark-gnome? The yum command downloads and installs the correct version automagically. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CCM, CME Integration
HI All, I got solved this issue. Thanks all for your help Arun On Sun, May 24, 2009 at 1:58 AM, David Backeberg dbackeb...@gmail.comwrote: On Wed, May 20, 2009 at 12:44 AM, Arun Kumar arunv...@gmail.com wrote: here is my problem: when I call from 6004 to my cme extension 4615, on 4615 I've configured noans timeout to 15 and then it goes to my unity express (cue) for voicemail so when I call my cme extension it rings for few seconds and then on my asterisk cli I see 500 Internal Server Error back from my CCM IP and getting standard asterisk message saying all circuits are busy now . as per my understanding it should go to my cue. You need to enable better debugging on the Cisco side. You shouldn't be getting a 500 internal server error. You need to debug the Cisco and find out what it says besides 500 internal server error. There should be logging for an error like that. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
On Sat, 23 May 2009, Steve Edwards wrote: On Sun, 24 May 2009, David @ULC wrote: Thanks a lot but which one to download before installing ? Which one what? Are you referring to sudo yum -y install wireshark wireshark-gnome? The yum command downloads and installs the correct version automagically. From asterisk@sedwards.com Sat May 23 21:16:37 2009 Date: Sat, 23 May 2009 21:16:37 -0700 (PDT) From: Steve Edwards asterisk@sedwards.com To: Steve Edwards asterisk@sedwards.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] HDD FULLL On Mon, 23 Feb 2009, Steve Edwards wrote: On Tue, 24 Feb 2009, David @ULC wrote: When I am trying to delete voice logs, [r...@vicidialnow monitor]# rm * -r -f -bash: /bin/rm: Argument list too long In the past 30 days, you've asked questions about configuring Apache to process PHP files, Vicidial, Ntework Cards, Auto Detecting hardware, BAT files on CentOS, Root Password not taking, How to find a file, Looking for a Free VOIP Billing and Soft Switch, What is a VPN, How do delete files, oh, and a couple of Asterisk questions. This is an Asterisk users list. We're here to help each other with Asterisk questions and problems, not to be your personal, for free, life coach. If you are being paid to work on an Asterisk system, you are in over your head. You are defrauding your boss and most likely will give him and everyone in the company a bad impression of Asterisk. Continuing to answer your questions will only continue to enable you. Please take a step back, buy some books, take some courses, practice on your own systems on your own time. I will continue to read your posts, but only for comic relief. I'm sorry. My ale-zheimer's must be kicking in. I'll try not to respond to your posts any more. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
