Re: [asterisk-users] Asterisk using 200% CPU and then crashing...

the same problems. Just that wasnt't related to installing any new hardware. You can check out the issue http://bugs.digium.com/view.php?id=10775 Could you provide your OS and glibc version? Also - can you try to disable IAX? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL

Re: [asterisk-users] Preflight check / lint

, however you should check syntax of applications, and priority ordering (if you are using it at all). Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Re: [asterisk-users] parse error in GosubIf

, I'd greatly appreciate your insight. This was really frustrating and is probably a stupid mistake. Try removing spaces around = Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Re: [asterisk-users] parse error in GosubIf

, this: Oct 17 22:14:54 -- Executing [EMAIL PROTECTED]:2] GosubIf(Zap/3-1, 1?notify|1) in new stack This means, the variable evaluates to 1 - only values are shown in log. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371

Re: [asterisk-users] PSTN failover

for those who use my system. Dialing twice like that without checking your return value is an invitation for future problems. Well, as far as i have tried - i never get ANSWERED in DIALSTATUS. Only thing that continues is h extension. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc

Re: [asterisk-users] CDR

://bugs.digium.com/view.php?id=10659). The thing is that one-channel CDRs without answer are not written. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?

pthread, and it should distribute load across multiple cores. However, i doubt that you will need that much for 35 simultenous calls. I have 8-core system that has web interface + sql + java + some other stuff running, and at 30 simultenous calls i get loadavg maximum of 3. Regards, Atis -- Atis

Re: [asterisk-users] Manager API ! (System) command

as asterisk user) If you really really want to do that, you can always use Originate manager action, and send it to System() app - but that's much more overhead, as that would create channel for every execution. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED

Re: [asterisk-users] Injecting a sound file into a bridged call

channels - one is bridged to Chanspy() and second to Playback(). I'm not sure what is the problem, but theoretically also this should work. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Re: [asterisk-users] Injecting a sound file into a bridged call

interrupting communications. However this will stop bridge of channels - so only one party will hear prompt, but second - silence. Regards, Atis - Original Message - From: Atis Lezdins [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 08, 2007 2:10 PM Subject

Re: [asterisk-users] Queue members, URI.

or SIP channel status. I would expect queue to show even Local channel as busy if there is active call trough it. I think this really can't be accomplished by dialplan logics, as dialplan is not executed upon show queues Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL

Re: [asterisk-users] Resolving digit strings using pound/hash.

for Read's and incoming calls. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] How to get asterisk to take a dump?

to asterisk and it would crash immediately to core - so you can play with it in gdb. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth

Re: [asterisk-users] IAXy and hook flash transfer

flag set in Dial() options? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Agent Callback Login in 1.4

. Is there any other way how i would get status indication in show queues? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation

Re: [asterisk-users] 3-way calling

/listinfo/asterisk-users -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth

Re: [asterisk-users] call relation in call transfer

-users -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation

Re: [asterisk-users] Asterisk Redundancy

way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype

Re: [asterisk-users] Asterisk Redundancy

. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth

Re: [asterisk-users] # to transfer calls

-- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided

Re: [asterisk-users] Asterisk crash and debug

received command 'Command' So, you're doing some CLI command trough AMI. I guess, it's show channels ;) I've seen it a lot on 1.2 (am i correct). I get rid of that o stopped only after upgrading to 1.4.10 Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype

Re: [asterisk-users] Asterisk and MS Exchange 2007

of direct SIP configuration, they don't seem complex. So, if you got the exchange, just play with it, and send us and microsoft the notes (in oo.org doc ;) Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800

Re: [asterisk-users] Authenticate() application and CDR

that moment. ResetCDR(w) would make you have two CDR records, one for each part (that can be linked together by using uniqueid). Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Re: [asterisk-users] asterisk crash and core dump: format_mp3.so

combination) in MOH directory - asterisk will pick up one with less translation. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now

Re: [asterisk-users] AgentCalbackLogin not loging in race condition ?

second, and is creating the trouble window. I changed that to allow 10 seconds of unavailability and the problem seems to be gone. -Carlos Shouldn't wrapuptime be used in this case? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371

Re: [asterisk-users] AgentCalbackLogin not loging in race condition ?

On Wednesday 19 September 2007 12:11:19 Carlos G Mendioroz wrote: Atis Lezdins @ 19/09/2007 06:05 -0300 dixit: On Wednesday 19 September 2007 11:43:39 Carlos G Mendioroz wrote: Previous mail did not go through. Following up... Carlos G Mendioroz @ 16/09/2007 13:27 -0300 dixit: Hi, I'm

Re: [asterisk-users] Queue serializes call delivery ?

expect a queue should work and in most cases, you will want to enable this new behavior. If you do not specify or comment out this option, it will default to no to keep backward compatability with the old behavior. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED

Re: [asterisk-users] How to cancel the password check in VoicemailMain()

, it is deprecated in 1.4, and should work at the end. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th

Re: [asterisk-users] asterisk crash and core dump

dumps. Please see http://www.voip-info.org/wiki-Asterisk+debugging Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon

Re: [asterisk-users] Call Center SoftPhone with Auto Answer

On 9/18/07, Joao Pereira [EMAIL PROTECTED] wrote: I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function,

<    1   2   3   4