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,

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: 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] 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] # 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 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] 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] 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] 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] 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] 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] 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] 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

) ... goto queue|2|1 } } context queue { 2 = { ResetCDR(w) Set(CDR(userfield)=queue) Queue(2); } } context agent { _X. = { Set(CDR(userfield)=agent) Dial(SIP/${EXTEN}); } } -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins

Re: [asterisk-users] Problem: features (from features.conf) not available if call was originated by manager API or call file

, and it works one way, but not another - then you should post a bug. You can also try using in your call file: Set: DYNAMIC_FEATURES=automon#... If this doesn't help either, you can dial to Local channel, and there execute a Dial(), and set variables if necessary. Regards, Atis -- Atis Lezdins

[asterisk-users] Play sound on hangup

Hi, Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? 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] Play sound on hangup

-- 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-- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Play sound on hangup

On Wednesday 17 October 2007 18:54:58 Philipp Kempgen wrote: Atis Lezdins wrote: Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? ---cut--- g- Proceed with dialplan execution at the current extension

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] Crash related to asterisk -rx ?

remote process unresponsive. There's a patch for 1.4, but i guess it wouldn't be hard to backport it for 1.2 http://bugs.digium.com/view.php?id=10847 you might also want the one mentioned in comments: http://bugs.digium.com/view.php?id=10888 Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs

Re: [asterisk-users] sip reload causes unreachable

time reloading the entire file. Is there a way to avoid this probleme or another way to add/remove sip phones dynamically ? Realtime? http://www.voip-info.org/wiki-Asterisk+RealTime+Sip Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell

Re: [asterisk-users] replace astdb with a cluster-capable sql database engine

mailinglist, it's been described numerous times. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth

Re: [asterisk-users] Silencing VoiceMail() app in * 1.4.10

should report a bug on http://bugs.digium.com , fixing this should be trivial. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Silencing VoiceMail() app in * 1.4.10

On 3/9/08, Godwin Stewart Horwich IT Services [EMAIL PROTECTED] wrote: On Sun, 9 Mar 2008 16:22:35 +0200, Atis Lezdins [EMAIL PROTECTED] wrote: I think that giving 's' argument should silence all prompts including auth-thankyou. You should report a bug on http://bugs.digium.com , fixing

[asterisk-users] Audiocodes MP124-FXS replying BUSY when line is not.

) - end of log --- -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Call tracing - Asterisk 1.4

that in DB, manipulating CDR is the way to go. When you will have more specific questions, please ask, i'm sure somebody will answer :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone

Re: [asterisk-users] Call tracing - Asterisk 1.4

of ${CDR(UNIQUEID)}, but you can use just ${UNIQUEID}. If you want to pass variable to child channels, you should make it inheritable. I'm using: Set(__call_id=${UNIQUEID}) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004

Re: [asterisk-users] queue log vs. cdr

=MEMORY select * from cdr where dst = 4010 and calldate between 2008030800 and 20080313145900 group by uniqueid; and then compare: SELECT * FROM a WHERE callid NOT IN (SELECT uniqueid FROM b) SELECT * FROM b WHERE uniqueid NOT IN (SELECT callid FROM a) Regards, Atis -- Atis Lezdins, VoIP

Re: [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?

nothing to worry about. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Order of queue member list

create_queue_member() function. This will allow you speed bonus from hashtable in some places, and will make sure the login time gets registred. You can also consider updating lastcall in set_member_paused() - i'm having both of those. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL

Re: [asterisk-users] Asterisk 1.4 reliability problems

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 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004

Re: [asterisk-users] ztdummy problem causing playback () to fail

application. If you have no intention to use it, you might very well remove. I've seen this problem once, however recompiling everything and restarting helped me. I would suggest you just doing make clean on zaptel and asterisk, then compile first zaptel, then asterisk. Regards, Atis -- Atis Lezdins

Re: [asterisk-users] asterisk.conf uniquename or sysname for uniqueid field in CDR

asterisk.conf without sysname and create shell script: #!/bin/bash cat asterisk.conf.template echo sysname=`hostname`. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Handling 3 different call ending causes

caller and callee hangups. I suppose dial time limit will match Callee hangup, but you can check that by ${ANSWEREDTIME} or some sort of timestamp checking before and after Dial (altough that would include ringing time) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL

Re: [asterisk-users] Handling 3 different call ending causes

On 3/20/08, Tobias Ahlander [EMAIL PROTECTED] wrote: Date: Wed, 19 Mar 2008 11:31:57 +0200 From: Atis Lezdins [EMAIL PROTECTED] Subject: Re: [asterisk-users] Handling 3 different call ending causes To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users

Re: [asterisk-users] hint status unavailable

, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] hint status unavailable

to allow combining of device states. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation

Re: [asterisk-users] callers in queue passed to agents who accept only one call at a time

Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] interrupting MOH

/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] interrupting MOH

the sound is done you start MoH back up again. Probably a bit more involved than what you want, but it dose work well for us. MATT--- On 4/2/08, Atis Lezdins [EMAIL PROTECTED] wrote: Sorry for top-posting, but seems everyone on this thread did so. Also that would be my suggestion for now

Re: [asterisk-users] Asterisk 1.4.19 crash with Realtime using SIP peers

type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1207763735 ipaddr: 192.168.1.123 regexten: cancallforward: yes setvar: call-limit: 4 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype

Re: [asterisk-users] question about queue

To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Global call limit

: this is needed for a PBX connectect with a poor ADSL having only 256kbit in upload: so i want permit only 3 calls (256 / 80 kbit) and rejecting the 4th. Any solutions? if (${GROUP_COUNT([EMAIL PROTECTED])}) function in combination with Set(GROUP(a)=x) or Set([EMAIL PROTECTED]) Regards, Atis -- Atis

Re: [asterisk-users] G729 license count...

* and not for the cards themselves. That's true. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth

[asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS

to call '[EMAIL PROTECTED]'. Giving up. -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS

on. Maybe You replied to wrong topic? Regards, Atos -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation

Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue

/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue

Atis Lezdins wrote: Queue will continue if called person hangs up (and there's no option). If caller hangs up, call goes to h extension in same context. Just the same way as Dial with 'g'. There's a change in 1.6 that allows called channel to continue if caller hangs up, so probably

Re: [asterisk-users] realtime queue callers

, but I intend to do that in future. It's been working stable on our production for several months. If You're interested, please reply, and I'll try to separate that patch out from other our patches. Currently I have it updated for 1.4.19, but also have some version for 1.4.14 Regards, Atis -- Atis

Re: [asterisk-users] realtime queue callers

On Tue, Apr 29, 2008 at 1:22 PM, Vieri [EMAIL PROTECTED] wrote: --- Atis Lezdins [EMAIL PROTECTED] wrote: On Mon, Apr 28, 2008 at 8:34 PM, Vieri [EMAIL PROTECTED] wrote: How can I get a list of the callers within a specific queue at any given moment? I need to get

Re: [asterisk-users] Remote host can't match request NOTIFY???

in sip.conf (mailbox= line). Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http

[asterisk-users] Realtime status feature - user feedback needed

a comment in bugtracker: http://bugs.digium.com/view.php?id=12556 Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] update DB on ringing/ catch ringing event

in future versions of Asterisk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] update DB on ringing/ catch ringing event

On Thu, May 8, 2008 at 12:34 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: Atis Lezdins schrieb: On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing

Re: [asterisk-users] Realtime status feature - user feedback needed

On Thu, May 8, 2008 at 1:07 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 07 May 2008 16:11:05 Atis Lezdins wrote: However I encountered a resistance from Asterisk developers, as they don't wish to accept my patches - because they don't wish to support another interface. As I

Re: [asterisk-users] Realtime status feature - user feedback needed

to give full overview of Asterisk Status. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth

Re: [asterisk-users] Newbie Queue: tricky problem with MOH

the call. Any thoughts? ___ -- 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 -- Atis Lezdins

Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?

channels. So, this should work with at least queue in ring-all mode (i feel that it would be correct if Dial would do that too) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1

Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?

). I suppose just a disconnect, because call was already bridged. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?

for configs, etc. Imagine what will happen if that one PSU will fail. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Polycom XML Files / asterisk

/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 -- Atis Lezdins, VoIP

Re: [asterisk-users] Dear asterisk-users@lists.digium.com May 87% 0FF

every message mentioning Microsoft :p Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth

[asterisk-users] Proposed changes for queue timeout

, and how I'm proposing. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] detecting which party hung up

will jump to h extension. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Camp / Callback feature in 1.4

-file. Of course you will need some additional handling in case if multiple callers decide to camp, or diferent protocols are used, etc. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689

Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)

it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)

On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial

Re: [asterisk-users] asterisk calls per second

+func+group Calls will still be received by asterisk, however you will be able to kick them off without proceeding with following dialplan logic. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800

Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)

On Thu, Jun 12, 2008 at 10:51 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup

Re: [asterisk-users] asterisk calls per second

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 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell

Re: [asterisk-users] Idiot's Question

-- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] cdr-custom/Master.csv rotation

/sh /usr/sbin/asterisk -rx 'logger reload' /dev/null 21 logger reload rotates logs. But not CSV . That's because the CSV CDR files are not held open. If they are not held open, you can can just move them away with mv, next CDR should just write new file. Regards A,tis -- Atis Lezdins, VoIP

Re: [asterisk-users] Agents getting stuck busy

. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] OT How Digium Saved My Bacon!

bug reporting, improvement suggestions, hell I debug and report on the entire new CDR/CEL branch :) ROFLno seriouslyI want one ;-) How about sending those out when certain amount of karma is reached? ;-) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL

Re: [asterisk-users] OT How Digium Saved My Bacon!

On Tue, Jun 17, 2008 at 10:03 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 2:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, I guess I am one of the lucky few to have one

Re: [asterisk-users] Reg call recording

duration). Mix and transcode (to some lower bandwidth codec) the rest of recordings at night time. Personally I record everything in ulaw, and either on listen or at night transcode to gsm for storage. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype

Re: [asterisk-users] CLI show queues NOT WORKING WELL

state_interface device when logging in agents. For more information please search for asterisk queue state, as this has been discussed several times. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1

Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!

with that? This fits perfectly for my needs. Is there a way how to exploit this? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!

show just duration and billsec (at least for 1.4), so i would defineately want this 1 second between 3 and 4 to show up in some record (preferrably in second CDR, as it's not talking time with first user anymore). Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED

Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?

here. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] QueueMemberStatus

. :) There is QUEUE_MEMBER_COUNT (in 1.4) and QUEUE_MEMBER (in 1.6) dialplan functions which allows to get count of members (in 1.6 also count of free / logged in members). You can use GetVar to evaluate that. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell

Re: [asterisk-users] finding out on hold channels

explaining how to do this by adding custom code to Asterisk sources, and I guess it could be already done in trunk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Customized Queuing Strategy

Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800

Re: [asterisk-users] skype and Asterisk opensource integration

to cell phones by cable, however it supports also skype (just 1 account). It will launch fake X server and original skype, and communicate with it. http://www.celliax.org/ Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371

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