Re: [asterisk-users] Customized Queuing Strategy

Alex Balashov a écrit : > Syed Nasruddin wrote: > > > >> 1. 10 Call Center Agents. >> >> 2. All the calls coming in will ALWAYS be routed to specific 5 agents, >> firstly. >> >> 4. IF ALL the first 5 agents are busy then ONLY then the call will be >> routed to next 5 Agents. >> > > Set

Re: [asterisk-users] Asterisk Queues problem

Syed Nasruddin a écrit : > > Hi, > > I have Asterisk 1.4.18 and I have been running call center queues on > it. Today it suddenly stopped adding inbound calls to queues. I am > facing with following error: _app_queue.c:3939 queue_exec: unable to > join queue “myqueue”_ > > In extension file: > >

Re: [asterisk-users] Whitepaper: How and to whom sell VoIP

As for me i mostly saw spellings mistakes, but that's me :) Grygoriy Dobrovolskyy a écrit : > i saw that billing iface somewhere else, maybe i am wrong... > > 2008/7/30 Mindaugas Kezys <[EMAIL PROTECTED] > > > Hello, > > Based on our own and our clients' experienc

Re: [asterisk-users] AMI able to call from known endpoint to unknown endpoint?

On Thu, Jul 31, 2008 at 05:28:42PM -0700, Stephen Cattaneo wrote: > Both are sitting behind a Linksys IP PBX (SPA9000). On the Linksys IP > PBX I have set the outside number 5000 to connect to 3001. 3002 does > not have a similar external mapping (this would defeat the purpose of > the test I am

Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

On Tue, Jul 29, 2008 at 01:48:26PM +1000, Lee, John (Sydney) wrote: > On the box, first of all, I just installed Zaptel 1.4.10.1. [..] > > BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] > Hi, just for (all of) you to know this is a known bug of zaptel < 1.4.11, the

Re: [asterisk-users] Cisco vs Asterisk

Al Baker a écrit : Quote "Yet amazingly (if this is, indeed, a source of amazement for you), CCM and other Cisco software can be just as buggy as anything OSS, if not worse. " This is simply NOT TRUE and shows a complete lack of understanding of modern software development. CISCO software i

Re: [asterisk-users] Cisco vs Asterisk

[EMAIL PROTECTED] a écrit : > Call me crazy, but why are you so keen on selling them an Asterisk box > when you don't even know if its capable of doing what you want to sell > it for? > I won't, i had the same felling ... > thats kinda scray actually. > Yep _

Re: [asterisk-users] Cisco vs Asterisk

voip crazy a écrit : > Hello all, > > A client of us, is thinking to migrate their actual PBX to a Cisco > CallManager. We want to sell him an asterisk box to complement the > Cisco PBX. > I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) > > Has asterisk all the functionali

Re: [asterisk-users] GotoIfTime Function

I would say you have two choices for that: opt 1, let the carrier provider do the ring and then answer, using Wait() or WaitForRing() opt 2, do it yourself using PlayTones() or Progess() broadband Voice a écrit : > Finally did it but only one more problem, I want it to ring onc

Re: [asterisk-users] iax2 trunk becomes unreachable (asterisk 1.4.21)

On Sat, Jun 21, 2008 at 05:30:51AM -0700, Vieri wrote: > Hi, > > I'm having trouble connecting two Asterisk boxes via a IAX2 friend trunk. > "iax2 show peers" on both boxes seem to show that all's fine (Status OK on > qualify=yes peer). > voip1 is an Asterisk 1.2.27 production server. > voip2 is

Re: [asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan

On Tue, Jun 17, 2008 at 10:56:06AM -0500, Kevin P. Fleming wrote: > Benoit Plessis wrote: > > Is it possible on a TE220p to deactivate the hardware echo canceler at > > will ? (With a function in the dialpan for example) > > example for fax SDA ,beeing able to shutdown th

[asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan

Is it possible on a TE220p to deactivate the hardware echo canceler at will ? (With a function in the dialpan for example) example for fax SDA ,beeing able to shutdown the echo canceler could give better results don't you think ? ___ -- Bandwidth and

Re: [asterisk-users] Euro_isdn PRI Line, callerid and usecallingpres"

Benoit Plessis a écrit : Hi, I'm having trouble with a TE220p PRI card and (outbond) caller identification. Previously with usecallingpres=no everything was Ok, one small difference between the BRI (B410p) was that the callerid needed to be stripped from one number. But then came the

[asterisk-users] Euro_isdn PRI Line, callerid and usecallingpres"

Hi, I'm having trouble with a TE220p PRI card and (outbond) caller identification. Previously with usecallingpres=no everything was Ok, one small difference between the BRI (B410p) was that the callerid needed to be stripped from one number. But then came the need to make "hidden" calls, and

Re: [asterisk-users] handling SIP trunk with limited concurent calls

Benoit Plessis a écrit : Gordon Henderson a écrit : On Thu, 5 Jun 2008, benoit plessis wrote: Hi, Now that we have a working asterisk server, i'm looking toward cost optimization :) We are actually testing a SIP provider, which has an interessting limitation: each account su

Re: [asterisk-users] Asterisk 1.6 vs 1.4?

Brent Davidson a écrit : ...I wonder why more vendors haven't adopted IAX yet? Well, even ZoIPer (ex IdeFisk) team, still recommend using SIP over IAX as "SIP is more mature and reliable in asterisk and zoiper", -- Benoit begin:vcard fn:Benoit Plessis n:Plessis;Benoit email;internet:[EMAIL PRO

Re: [asterisk-users] handling SIP trunk with limited concurent calls

Gordon Henderson a écrit : > On Thu, 5 Jun 2008, benoit plessis wrote: > > >> Hi, >> >> Now that we have a working asterisk server, i'm looking >> toward cost optimization :) >> >> We are actually testing a SIP provider, which has an interessti

[asterisk-users] handling SIP trunk with limited concurent calls

Hi, Now that we have a working asterisk server, i'm looking toward cost optimization :) We are actually testing a SIP provider, which has an interessting limitation: each account support at max only two concurrent calls. My problem is how to combine multiple accounts and fail back to PSTN lines

Re: [asterisk-users] Asterisk on iPhone

You might be looking for that instead http://sip.free.fr/index.html.en Andrea Cristofanini a écrit : > Hi > I just saw this now ! > does the microphone and speaker works ? > Can you use it like softphone for recive calls ? > Regards Andrea > C F ha scritto: > __

Re: [asterisk-users] Not hearing first prompts

d card and phone using another one. Anyway instead of doing a Wait(), i used a Anwser() + Playback(silence/1) to get around this kind of thing. Since it's most probably problem while decoding first bunch of audio sample, using Wait() won't help. -- Benoit Plessis

Re: [asterisk-users] queue problem

On Wed, May 14, 2008 at 01:27:38AM +0800, Rilawich Ango wrote: > I have a queue with the following setting. > total queue member =30, autofill=1, timeout=25, monitor_format=wav49 > asterisk 1.4.18 > In busy hour, the loading of CPU reaches over 300%. At that moment, > all members are occupied and

Re: [asterisk-users] Problem with SIP Subscription Status

our version. Did you look through the changelog / bugs tracker to see if your problem has already been reported ? -- Benoit Plessis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

w.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+IAX+agent Or if you want your queue agent to be joinable with internal calls while in communication with a queue member. -- Benoit Plessis +33 6 77 42 78 32 <[EMAIL PROTECTED]>

Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

> http://www.asterisk.org/node/48360 > -- Benoit Plessis +33 6 77 42 78 32 <[EMAIL PROTECTED]> +33 4 67 28 06 96 ___ -- Bandwidth and Colocation Provided by http://www.api-d

Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

Russell Bryant a écrit : > I don't see why this wouldn't help. You just list the IAX2 peer as the device > Asterisk uses to determine the state of the agent. > Well i've read elsewhere that only SIP peers did support the use flag ? -- Benoit Plessis

Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

d-slash in the device name, and returns if it > doesn't find one: Why not modifying func_devstate instead to support Custom/ ? btw why is it limited to some channel name ? it would be nice to use the Agent/xxx channel directly -- Benoit Plessis+

Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

Benoit Plessis a écrit : > Russell Bryant a écrit : > >> Alternatively, if you would like to control the usability of an agent >> through >> the dialplan, then you could use the DEVICE_STATE() function to >> create a custom >> device state. Then, you cou

Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()

Russell Bryant a écrit : > Benoit Plessis wrote: > >> So i'm wondering if someone already as made a dialplan function that >> could toggle the 'Use' flag of >> an agent ? or if this kind of function would be integrated into the core >> if i build

[asterisk-users] app_queue New Function ToggleQueueMemberUse()

Hi, I'm using a Queue in asterisk with IAX2 peers and my agents are also doing outbond calls. Actually we are using a GROUP() function like that to prevent users from beeing Dialed while in communication: exten => s,1,gotoif($[${GROUP_COUNT([EMAIL PROTECTED])}=0]?:busy) exten => s,n,Set([EMAIL

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

to do things, why adding another one, especialy if it's for caching reasons. While we cannot say that asterisk fall into the KISS rule, it's not a reason to let it grow. -- Benoit Plessis +33 6 77 42 78 32 <[EMAIL PROTECTED]>

Re: [asterisk-users] mISDN on Debian Lenny

Philipp Kempgen a écrit : > Benoit Plessis wrote: > > >> Well i tried a debian/lenny with an mISDN patched for 2.6.24 >> > > Are those patches available somewhere? Pointers? > > Regards, > Philipp Kempgen > > It's a patch i got from

Re: [asterisk-users] Asterisk in Production ?

Tzafrir Cohen a écrit : > On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote: > >> Tzafrir Cohen a écrit : >> >>> On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: >>> >>> >>> >>>> Her

Re: [asterisk-users] Asterisk in Production ?

Tzafrir Cohen a écrit : > On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: > > >> Here it is, but since the AsteriskNow release has stripped the binary >> i fear it won't be of much use: >> > > Is there any "-debug" package f

Re: [asterisk-users] Asterisk in Production ?

Steve Totaro a écrit : > On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote: > >> lordfuknowsyou a écrit : >> >> >>> Vinícius Fontes wrote: >>> >> > >> > I use 1.4.18 with no problems. We have

Re: [asterisk-users] Asterisk in Production ?

Tilghman Lesher a écrit : > On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote: > >> Tilghman Lesher a écrit : >> >>> On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: >>> >>>> lordfuknowsyou a écrit : >>>> &

Re: [asterisk-users] Asterisk in Production ?

Tilghman Lesher a écrit : > On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: > >> lordfuknowsyou a écrit : >> >>> Vinícius Fontes wrote: >>> >>> I use 1.4.18 with no problems. We have quite a few users(125 total >>> between branche

Re: [asterisk-users] Asterisk in Production ?

Steve Totaro a écrit : > On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <[EMAIL PROTECTED]> wrote: > >> Any IAX2 phone or mostly SIP ? >> Do you use Call Queues ? >> >> We have less user than that, less concurrent call but quite a few >> crash/deadl

Re: [asterisk-users] Asterisk in Production ?

lordfuknowsyou a écrit : > Vinícius Fontes wrote: > > I use 1.4.18 with no problems. We have quite a few users(125 total > between branches), but the call volume at the most has been around 15 > active calls at a time. > Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less

[asterisk-users] Asterisk in Production ?

Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some "Avoiding I

[asterisk-users] Passing Dial URL argument through Asterisk

Hi, I'm able to send an HTML Frame to an IAX2 phone like Zoiper, Using Dial(IAX2/...,,,"http://site/uri";) but now i need to do this Inbound Call == Zap ==> Asterisk-1 [main diaplan, build the needed Url ] == IAX2 ==> Asterisk/2 == IAX2 ==> Phone By using a tcp trace i can see that the HT

Re: [asterisk-users] Strange CLI behaviour

lokotes2 a écrit : > Hi, > I'm using Asterisk 1.4.17 and 1.4.19 versions, some time ago I've > noticed that cli command 'core show channels' does not show all data. > It returns only header or one line of data. > After that, auto completition of commands (hitting TAB) freezes cli... > Does anybody