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 up two queues. Call Queue() on the first queue - corresponding to #1 - with a rather strict timeout. Fall back on the second queue. More sophisticated strategies require either the modification of the source code for app_queue, or custom queue implementation in AGI Isn't this possible with agent priority ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 attempting). [...] Is it possible (and if yes, how can I do this) to use the AMI's originate to call from 5000 to 3001? Don't you mean to 3002 ? either that or you will make a loopback call ... Anyway, if that's what i understood it's impossible. You can see a PABX a little bit little a network NAT device (well in your specific problem). Behind the Linksys there is a private network, with one public adresse, which is a static map to one private adresse. Well at least it's how your asterisk IPBX will see things. Basically when using asterisk to connect (call) two extension, asterisk will dial each extension, and then connect them. Since he doesn't have any access to your 3002 internal extension, there is no way this could work. When calling from 3001 you can reach the 3002 extension, but only because you are within the same network, to keep the metaphor -- Benoit ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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] mailto:[EMAIL PROTECTED] Hello, Based on our own and our clients' experience we compiled short manual: How and to whom sell VoIP Hope it can be useful to some of you also. You can download it from our site: http://www.kolmisoft.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It’s a production machine. Thanks Syed nasr I would recommend upgrading your asterisk to at least 14.20.1 I have had many troubles with queues, SIP and IAX with asterisk 1.4.18 that have been fixed in the following releases ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 firmware upload procedure is taking some time, operating like a freeze during the process, so this message appears. But this isn't a real problem, as it doesn't have any consequences appart from the message. -- benoit ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 is developed in a CMM environment. It has a formal test methodology and uses Automated Testing on EACH new release to ensure that 100% of the software that functioned in the Last Release, actually works in this release. So every professional software is now free of bug ?? And every professional team that build thoses software are perfect people ??? And every automated process made to validate thoses software are so complete that nothing is left out Well that doesn't explain the vast quantity of bug fixed in every Cisco IOS release, or even the vastly unfinished CRS-1 product from ... cisco... Further, there is mandatory soak-testing for all new software. Sorry, anyone who wants to compare Professional TELCO GRADE software development with Open Source is just Completely and Totally freakin clueless. Yes, so professional telco grade doesn't have bug fix release ?? -- Benoit begin:vcard fn:Benoit Plessis n:Plessis;Benoit email;internet:[EMAIL PROTECTED] tel;home:+33 9 52 49 25 06 tel;cell:+33 6 77 42 78 32 x-mozilla-html:FALSE version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the Cisco Call Manager funcionalities? To answer your questions, one would need to know what exactly are all the functionalities of a Cisco Unity server, and more specificaly, what are the needs of your client. But i'm pretty sure the voip-info wiki can answer the asterisk part... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 once before going to the context or playing the background message. [day_menu] exten = s,1,Answer() exten = s,2,Background(welcome-message) exten = s,3,Dial(SIP/5960,200,rt) ; week day goes to Philadelphia Office [weekend__menu] exten = s,1,Answer() exten = s,2,Background(welcome-message) exten = s,3,Dial(SIP/5961,200,rt) ; weekend goes to Delaware Office [night_menu] exten = s,1,Answer() exten = s,2,Background(officeclosed) exten = s,3,Hangup ; ;incoming exten = 1866x,1,GotoIfTime(8:00-18:00|mon-sun|*|*?day_menu,s,1) exten = 1866x,n,Goto(night_menu,s,1) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 an Asterisk 1.4.21 experimental server in the same gigabit LAN. Have you tried another 1.4.xx version ? There is a few problems introduced by this release and i already experienced this exact problem. -- Benoit ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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] 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 the echo canceler could give better results don't you think ? All echo cancelers using Zaptel/DAHDI already disable themselves when FAX or modem communications are used, based on reception and detection of the CED tone that FAX machines and modems generate to make that happen. You can tell this happened by looking at the channel in Asterisk using 'zap show channel' or 'dahdi show channel' as it will show you that the echo canceler was disabled automatically. Karamba ! It mean than that my last option is to put all three digium cards in one box and hope, that it'll fit and that it'll work better :( (actually i'm using one server with two cards (TE220, B410) which is linked with an (slinear) IAX peer to another server with one TDM800 to talk to the fax machine. -- Benoit ___ -- 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] 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 so to enable usecallingpres and SetCallerPres(). if running SetCallerPres(prohib) then the end call get 'private number' which is what we want but with SetCallerPres(allowed) or SetCallerPres(allowed_not_screened) i'm not able to get the number i want, i got the line identification number in any case (even with a callerid of 10 digits) any idea ? my zapata.conf: 8 [trunkgroups] [channels] context=from-rnis-t2-dys language=fr switchtype=euroisdn signalling=pri_cpe callwaiting=no threewaycalling=no callprogress=no busydetect=no pridialplan=unknown prilocaldialplan=dynamic priindication=outofband internationalprefix=00 nationalprefix=33 localprefix= privateprefix= unknownprefix= relaxdtmf=yes hidecallerid=no usecallingpres=yes echocancel=yes faxdetect=incoming immediate=no group=1 channel = 1-15 8 ___ -- 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] 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 need to make hidden calls, and so to enable usecallingpres and SetCallerPres(). ... Well, it was solved by changing prilocaldialplan=dynamic to prilocaldialplan=unknown Is there some good technical documentation about PRI lines somewhere ? there is a lot of voodoo magic in configuring thoses lines with asterisk ... -- Benoit begin:vcard fn:Benoit Plessis n:Plessis;Benoit email;internet:[EMAIL PROTECTED] tel;home:+33 9 52 49 25 06 tel;cell:+33 6 77 42 78 32 x-mozilla-html:FALSE version:2.1 end:vcard ___ -- 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] 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 if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
--
Benoit
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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 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 if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($[${DIALSTATUS} = CONGESTION]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION ... (You
may not need the call-limit=2, if they check for you, then if at a later
date, they increase the limit, then you don't need to change anything)
Gordon
Isn't there a risk of getting a CONGESTION message from the other party ?
benoit
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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 PROTECTED] tel;home:+33 9 52 49 25 06 tel;cell:+33 6 77 42 78 32 x-mozilla-html:FALSE version:2.1 end:vcard ___ -- 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] 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 support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($[${DIALSTATUS} = CONGESTION]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION ... (You
may not need the call-limit=2, if they check for you, then if at a later
date, they increase the limit, then you don't need to change anything)
Gordon
Isn't there a risk of getting a CONGESTION message from the other party ?
benoit
Another problem i foresee is long delay in dialing sequence when
asterisk will have to dial using 4/5 account
before having a working channel
i think i should look after another sip provider
--
Benoit
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email;internet:[EMAIL PROTECTED]
tel;home:+33 9 52 49 25 06
tel;cell:+33 6 77 42 78 32
x-mozilla-html:FALSE
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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: ___ -- 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] Not hearing first prompts
Alan Lord wrote: Unfortunately not. It is the same if we use our Siemens DECT/SIP handsets or the Ekiga softphone... I recall having this problem once before and that it went away when I changed from Ekiga to Twinkle. When I get chance, I will re-install Twinkle and see if that exhibits the same problem. I've always had this kind of problems with ekiga, the first 2/3 seconds of sound are of bad quality (cutted, metallic, ..) and always missing one or two seconds, and this last for at least 3 years. this remind me that i still hasn't verified if this as been reported as a bug to ekiga team it is the reason why i'm using twinkle as a phone client, and twinkle isn't a good client (weird translation, usability inexistant) but at least audio is good and it can ring on a sound 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 +33 6 77 42 78 32 [EMAIL PROTECTED] +33 4 67 28 06 96 ___ -- 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 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 many calls are waiting in the queue. There will be choppy and line cut at such high CPU loading. Hi, I was having huge problems with AsteriskNow 1.0.1 which is packaged with asterisk 1.4.18(.1? not sure). Most of them came with the deploiment of our support center call queue. With only 2/3 agents and max 6/10 simultaneous calls the system goes wazaa and eat the 4 cores of the xeon 1.6 cpu, users gets stucks in 'in use' state (while normaly this flags doesn't work) and everything goes from bad to worse. I've rebuild it using etch/amd64 and manual build of asterisk 1.4.20rc2 (1.4.19.2 wasn't out and i have some IAX calls) and now everything is fine. My questions: 1. What is the max capacity of a server to handle a queue in term of queue member and calls? Do you use IAX2 ? there is major improvement in this with 1.4.19.2 / 1.4.20. 2. After every 25s, the call will be switched from agent to another agent. Can I do something, say execute a CLI or shell command before it switches to another agent? -- Benoit ___ -- 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] Problem with SIP Subscription Status
Hach Segal a écrit : Hello All, I've been having some intermittent trouble with an Asterisk 1.2.10 Before anything else did you tried an updated asterisk 1.2 The last one is 1.2.28 or something like that, and there has been a lot of security patches, and fixes since your 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 or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()
Patrick a écrit : Can you please tell me where you found the backport? Thanks, Patrick Russel already sent the links: Yes, this is also from 1.6, but an unsupported backport of DEVICE_STATE(), exists, as well. http://www.asterisk.org/node/48325 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-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_queue New Function ToggleQueueMemberUse()
Steve Totaro a écrit : Depending on the phone, the simplest and easiest way to handle this is to set the phone to not accept call waiting. That's not always possible, if you use IAX2 then that's not even an option (at least on the PBX side): http://www.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] +33 4 67 28 06 96 ___ -- 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] 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 PROTECTED])
...
But this system fill the asterisk console with queue dial attempt, and
even with our small system
it make debugging a pain in the a**.
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 it ?
Regards,
benoit
CCed to asterisk-dev but i'm not subscribed so please make me in copy.
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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 it ? This is a slightly different approach, but have you seen the state interface code that is in Asterisk 1.6? There is a backport of the code for 1.4 floating around somewhere, I think. It allows you to specify a different device for a queue member that app_queue will use to determine the state of an agent. So, you can still list a Local channel for dialing, but Asterisk will look at the state of SIP/myphone, for example, to know whether the agent is busy or not. ok but since we are using IAX2 with ZoIPer for sendurl() handling this won't help 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 could list your custom device as what app_queue should look at before attempting to call the agent This is more interesting :) Is it from 1.6 too ? ___ -- 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] 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 could list your custom device as what app_queue should look at before attempting to call the agent This is more interesting :) Is it from 1.6 too ? don't bother, i found the backport ___ -- 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] app_queue New Function ToggleQueueMemberUse()
On Fri, May 09, 2008 at 10:49:02AM -0400, Watkins, Bradley wrote: Russell Bryant wrote: One problem with that cunning plan is that using custom device states doesn't work. The code for handling device state changes in app_queue is looking for a forward-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+33 4 67 36 42 59 [EMAIL PROTECTED] ___ -- 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] 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 +33 6 77 42 78 32 [EMAIL PROTECTED] +33 4 67 28 06 96 ___ -- 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 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: 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 for asterisknow's asterisk package? On RedHat they are generated automatically. On Debian they require some extra settings, and has been present in recent Asterisk packages (the asterisk-dbg package) but not in all of the smaller modules packages. Nope, already tried this before posting but nothing like that appears on conary I looked again at http://rbuilder.rpath.com/ and searched for the package asterisk. It does seem to have a subpackage called asterisk:debuginfo. I'm not able to install it but i'll look further, conary is a tricky software to say the least anyway, i'll be migrating on a debian asap, since i now this much better and the advantages of AsteriskNow keep reducing Off topic: That is not to say you should not try Debian ASAP ;-) Well i tried a debian/lenny with an mISDN patched for 2.6.24 but it lead to kernel panic / server reboot after 4/5 calls on the B410p. No problem on the T220b but i need both cards ... I think i'll have to reinstall an debian/etch and either try the packaged asterisk 1.2 or manually build an 1.4 + zaptel + misdn. Everything i was looking away from when i initially choosed asteriskNow To help you with that, here's a live CD: http://updates.xorcom.com/iso/ ___ -- 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] 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 the gentoo portage site, should be made of some mISDN commit in the git tree. but I don't recommend using them, i got two kernel panic and a hard reboot after 4/5 calls http://kambing.ui.edu/gentoo-portage/net-dialup/misdn/files/misdn-2.6.24.diff ___ -- 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] Realtime status feature - user feedback needed
Tilghman Lesher a écrit : On Wednesday 07 May 2008 17:27:33 Atis Lezdins wrote: So all together - I'm saying there could be really simple interface for all this - no troubles with locking of lists or keeping persistent connections. Why would user application need to take care of all this, if DB engine can do that. Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. I second that, If there is already a way 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] +33 4 67 28 06 96 ___ -- 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 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 user than that, less concurrent call but quite a few crash/deadlocks ___ -- 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 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/deadlock Try SIP only if you can and report back. I think you will confirm what is pretty much a silent consensus (even among Digium Devs). Hi, that's what i was planning seeing all thoses answers. We initialy choosed IAX2 for the sendurl() support but i'll set-up a test periode in SIP-only to compare. Thanks, Steve Totaro Thanks to 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
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 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 user than that, less concurrent call but quite a few crash/deadlocks Have you reported these issues on the bugtracker? Well, the problem is finding usefull data to report. I've 4 core dumps thats show differents things: two seems to be related to ControlPlayback: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 #1 0x0809c579 in ast_readframe () #2 0x0809defc in ast_streamfile () #3 0x0805e786 in ast_control_streamfile () #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #5 0x08298700 in ?? () #6 0xb470aec0 in ?? () #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #9 0x in ?? () One is pretty generic: #0 0x0809c9bc in ast_closestream () #1 0x08085d91 in ast_hangup () #2 0x080cd3d8 in pbx_builtin_setvar_helper () #3 0x080cf08e in ast_pbx_outgoing_exten () #4 0x080fde65 in ast_inet_ntoa () #5 0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0 #6 0xb703667e in clone () from /lib/tls/libc.so.6 and the latest is thread/iax2 related: #0 0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0 #1 0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #2 0x0079 in ?? () #3 0x in ?? () #4 0xb547a148 in ?? () #5 0x080f0508 in ast_sched_add_variable () #6 0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #7 0x0012 in ?? () But my main problem is when the system just froze, it start mostly by the Queue not working anymore, with member stuck in 'in use' stack (should not happen with IAX2 agent IIRC, given that we had to build macros using GROUP() to detect in use IAX2 agent) Then the console (asterisk -rcTvvv) start to freeze (completion doesn't work, message stop from being displayed and even command output is lost). And i'm reading http://www.asterisk.org/developers/bug-guidelines which speak of using SVN trunk version of asterisk, thing i'm not really eager to try on a live system... ___ -- 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 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 : 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 user than that, less concurrent call but quite a few crash/deadlocks Have you reported these issues on the bugtracker? Well, the problem is finding usefull data to report. I've 4 core dumps thats show differents things: two seems to be related to ControlPlayback: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 #1 0x0809c579 in ast_readframe () #2 0x0809defc in ast_streamfile () #3 0x0805e786 in ast_control_streamfile () #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #5 0x08298700 in ?? () #6 0xb470aec0 in ?? () #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #9 0x in ?? () I'd love to see a 'bt full' on this one. Here it is, but since the AsteriskNow release has stripped the binary i fear it won't be of much use: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 No symbol table info available. #1 0x0809c579 in ast_readframe () No symbol table info available. #2 0x0809defc in ast_streamfile () No symbol table info available. #3 0x0805e786 in ast_control_streamfile () No symbol table info available. #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #5 0x08298700 in ?? () No symbol table info available. #6 0xb470aec0 in ?? () No symbol table info available. #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #9 0x in ?? () No symbol table info available. #10 0x in ?? () No symbol table info available. #11 0x in ?? () No symbol table info available. #12 0x0bb8 in ?? () No symbol table info available. #13 0x2f727669 in ?? () No symbol table info available. #14 0x65696c63 in ?? () No symbol table info available. #15 0x2f73746e in ?? () No symbol table info available. #16 0x6a6e6f62 in ?? () No symbol table info available. #17 0x2d72756f in ?? () No symbol table info available. #18 0x6e656962 in ?? () No symbol table info available. #19 0x756e6576 in ?? () No symbol table info available. #20 0x6568632d in ?? () No symbol table info available. #21 0x6f702d7a in ?? () No symbol table info available. #22 0x62726577 in ?? () No symbol table info available. #23 0x6974756f in ?? () No symbol table info available. #24 0x2d657571 in ?? () No symbol table info available. #25 0x76726573 in ?? () No symbol table info available. #26 0x73656369 in ?? () No symbol table info available. #27 0x696c632d in ?? () No symbol table info available. #28 0x00746e65 in ?? () No symbol table info available. #29 0x0001 in ?? () No symbol table info available. #30 0xb470af20 in ?? () No symbol table info available. #31 0x081aa084 in ?? () No symbol table info available. #32 0x001b in ?? () No symbol table info available. #33 0x0025 in ?? () No symbol table info available. #34 0x0028 in ?? () No symbol table info available. #35 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #36 0x in ?? () No symbol table info available. #37 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #38 0x0829c4a8 in ?? () No symbol table info available. #39 0x0bb8 in ?? () No symbol table info available. #40 0x in ?? () No symbol table info available. #41 0xb470aec0 in ?? () No symbol table info available. #42 0x in ?? () No symbol table info available. #43 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #44 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #45 0x in ?? () No symbol table info available. #46 0x in ?? () No symbol table info available. #47 0x in ?? () No symbol table info available. #48 0x in ?? () No symbol table info available. #49 0x08298700 in ?? () No symbol table info available. #50 0xb705b631 in strcasecmp () from /lib/tls/libc.so.6 No symbol table info available. #51 0x080c8740 in pbx_substitute_variables_helper () No symbol table info available. #52 0x080cd170 in pbx_builtin_setvar_helper () No symbol table info available. #53 0x080cf08e in ast_pbx_outgoing_exten () No symbol
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 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 user than that, less concurrent call but quite a few crash/deadlocks Try SIP only if you can and report back. I think you will confirm what is pretty much a silent consensus (even among Digium Devs). Thanks, Steve Totaro ___ -- 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 I've tried SIP only but i already got one 'stuck' Queue member: Members: Local/[EMAIL PROTECTED] with penalty 10 (dynamic) (In use) has taken 1 calls (last was 45 secs ago) Local/[EMAIL PROTECTED] with penalty 20 (dynamic) (Not in use) has taken no calls yet Callers: 1. Zap/10-1 (wait: 0:18, prio: 0) [May 6 17:48:35] NOTICE[2047]: app_queue.c:2152 wait_for_answer: No one is answering queue 'support' (1/0/0) asterix*CLI core show channels Channel Location State Application(Data) SIP/rtournier-081ef2 (None) Up Bridged Call(Local/[EMAIL PROTECTED] but the other end of the bridged call is long gone ___ -- 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 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 for asterisknow's asterisk package? On RedHat they are generated automatically. On Debian they require some extra settings, and has been present in recent Asterisk packages (the asterisk-dbg package) but not in all of the smaller modules packages. Nope, already tried this before posting but nothing like that appears on conary anyway, i'll be migrating on a debian asap, since i now this much better and the advantages of AsteriskNow keep reducing as a matter of fact i already now that some thing that doesn't work under AstNow (my siemens sip hardphones, and my SIP provider (Keyyo) at least) work with the debian packaged asterisk. Well for the sip provider it's not that it doesn't work, more than the only way to have some sound is to use the 'm' flag of the Dial() command to have the moh played during the ringing. Given that, i got some sound when the call is established ... -- Benoit ___ -- 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] 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 HTML frame is send to Asterisk/2
but there is no ${URL} variable set
and so i don't know how to transfer it to the queue and then to the phone.
Any idea ?
Did asterisk even parse the inbound HTML frame ? i can't find evidence
of this in the source code (rapid glance).
Regards,
___
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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 has the same problem? regards, Lokotes. Do you use any zaptel cards ? We have two cards in our Asterisk server (B410p (misdn) and TE220p (zapata)) and this kind of problem seem to happen since the day we started to use the TE220p (auto completion freeze and then the console start to fail big time, no more logging message and then even restart command doesn't work). regards ___ -- 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
