Re: [asterisk-users] Pickup(), PickupChan()... PickupQueue()? (Niccol? Belli)
If there is no phone ringing (because all operators are busy and the call is still waiting in the queue) then I cannot pickup the call with Pickup(Queuename@PICKUPMARK): app_directed_pickup.c:302 pickup_exec: No target channel found for magazzino@PICKUPMARK. Any idea? Niccolo' On lunedì 31 ottobre 2016 20:16:30 CET, Freddi Hansen wrote: Hi, I'm currently using Pickup() to pickup calls from queues, but this is VERY annoying because often users from different queues dialed the very same extension (for example they pressed '1' to reach something but in different submenus). Other times they didn't dial anything but they end up in the very same queue, so the extension to pickup is the number they called. So every time I want to send users to a queue I have to put a Goto() before the Queue() app because I need to uniquely identify the extension (for example Goto(QueueName,1)). This is annoying. Really annoying. It also makes the dialplan hard to read. Since we also have PickupChan() is to would be nice to have PickupQueue() too. That way we shouldn't care about the extension, we should simply write PickupQueue(QueueName). Simple and clear, the dialplan thanks. Hi, you could use the PICKUPMARK with the Pickup(). before you call the Queue app you set PICKUPMARK=Queuename. When you want to pickup the call you do Pickup(Queuename@PICKUPMARK) to only get calls in the Queue with Queuename. b.r. Freddi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup(), PickupChan()... PickupQueue()? (Niccol? Belli)
On lunedì 31 ottobre 2016 20:16:30 CET, Freddi Hansen wrote: you could use the PICKUPMARK with the Pickup(). before you call the Queue app you set PICKUPMARK=Queuename. When you want to pickup the call you do Pickup(Queuename@PICKUPMARK) to only get calls in the Queue with Queuename. That sounds good, I will try it. Thanks, Niccolo' -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup(), PickupChan()... PickupQueue()?
Hi, I'm currently using Pickup() to pickup calls from queues, but this is VERY annoying because often users from different queues dialed the very same extension (for example they pressed '1' to reach something but in different submenus). Other times they didn't dial anything but they end up in the very same queue, so the extension to pickup is the number they called. So every time I want to send users to a queue I have to put a Goto() before the Queue() app because I need to uniquely identify the extension (for example Goto(QueueName,1)). This is annoying. Really annoying. It also makes the dialplan hard to read. Since we also have PickupChan() is to would be nice to have PickupQueue() too. That way we shouldn't care about the extension, we should simply write PickupQueue(QueueName). Simple and clear, the dialplan thanks. Best regards, Niccolo' Belli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hangout + cid callback + dtmf dial
Hi, I think almost everybody did it once: you call the pbx, the pbx hangs out, it calls back using the caller id, you type the number you want to dial using dtmf and the pbx calls the number for you. Such a way you can make a completely free call. Of course you should implement CLI whitelists and passwords to enhance security. Unfortunately I don't use this function since several years, I lost the code and I don't really remember how I achieved it. Do someone has some code to share? Niccolò -- www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending SMS with a Portech MV-374 GSM Gateway
Hi, I have a Portech MV-374 GSM Gateway and I'd like to send SMS from a web page to confirm the subscriptions. How can I achieve it? Is Asterisk of any use to send SMS with the Portech? I really have no idea because I know nothing about the whole SMS thing... Thanks, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agents in more than one queue at once
Il 22/10/2012 18:44, Alex Forster ha scritto: *DEVELOPERS* - If I took a crack at fixing this issue, what general tips do you have for me to make it most likely that my solution can be integrated into HEAD? I believe I can justify spending some time at work to deal with this, but not without at least a decent chance that the work will be integrated into mainline (assuming it doesn't suck, of course:) Nice to hear you are willing to work on it. I suggest you to ask on asterisk-dev ;) Cheers, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk sends wrong fxs 'Idle' hints
It seems fixing this issue and [1] would require significant efforts: https://issues.asterisk.org/jira/browse/ASTERISK-20556 Too bad, it means DAHDI is a no way for me, I will have to switch to SIP DECTs :( Niccolò [1]http://asteriskfaqs.org/2012/10/08/asterisk-users/how-to-avoid-automatic-answer-with-callwaitingyes-on-fxs-channels.html Il 09/10/2012 13:34, Niccolò Belli ha scritto: Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the Idle state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him because the light on their function keys switched off! Cheers, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to avoid automatic answer with callwaiting=yes on fxs channels?
It seems fixing this issue and [1] would require significant efforts: https://issues.asterisk.org/jira/browse/ASTERISK-20556 Too bad, it means DAHDI is a no way for me, I will have to switch to SIP DECTs :( Niccolò [1]http://asteriskfaqs.org/2012/10/09/asterisk-users/asterisk-sends-wrong-fxs-idle-hints.html Il 08/10/2012 13:20, Niccolò Belli ha scritto: I will make an example: A is an fxs phone with callwaiting=yes in chan_dahdi.conf X calls A. A answers. Y calls A. A hears the call waiting tone. Now if A hangs up before X, then A rings again (which is what I want). BUT if X hangs up first, then A automatically answers Y without even ringing. Is there a way to avoid it? If X hangs up first I want A to hear the busy tone until it hangs up too. Then I want A to ring again. Otherwise if both A and X hang up at the same time there is no way to know what happened. Thanks, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk sends wrong fxs 'Idle' hints
Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the Idle state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him because the light on their function keys switched off! Cheers, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk sends wrong fxs 'Idle' hints
Il 09/10/2012 13:34, Niccolò Belli ha scritto: Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the Idle state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him because the light on their function keys switched off! Cheers, Niccolò I made a video of the bug: http://files.linuxsystems.it/files/dahdi_hints_bug.webm Can someone help me? Thanks, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover router recommendation
Il 09.10.2012 21:24 Mike Diehl ha scritto: I hope no one considers this off topic... I have a phone customer who wants 2 Internet connections so that if one goes down, he can use the other for phone service. So, I'd like to get a recommendation for a relatively inexpensive router that can perform this function. Also, when the failover occurs, the phone's IP address will obviously change. So, how can/should I configure this to minimize my customer's down-time? http://www.traverse.com.au/geos21-dual-adsl2-x86-router-appliance I achieved fallback in less than 10 seconds flushing routing cache and nat tables with nearly zero false positives (I can do even better but I prefer having less false disconnections). I don't use this router but a Traverse Solos PCI Adsl2+ card and a linux box. Cheers, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover router recommendation
Il 09.10.2012 23:04 James Sharp ha scritto: Do you have your phones set for a short register time? Otherwise the far end might have stale contact information to send incoming calls back to. Actually I use the failover only for the nat clients, my pbx has a public ip on the interface and it receives the incoming calls from PRI (which I use as outgoing fallback too). externaddr should be another thing you should take care of. Let me know how you will work around such things, my main focus had been nat clients and I did just a few tests with asterisk. Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to avoid automatic answer with callwaiting=yes on fxs channels?
I will make an example: A is an fxs phone with callwaiting=yes in chan_dahdi.conf X calls A. A answers. Y calls A. A hears the call waiting tone. Now if A hangs up before X, then A rings again (which is what I want). BUT if X hangs up first, then A automatically answers Y without even ringing. Is there a way to avoid it? If X hangs up first I want A to hear the busy tone until it hangs up too. Then I want A to ring again. Otherwise if both A and X hang up at the same time there is no way to know what happened. Thanks, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove the call waiting tone without disabling
Hi, Il 02/10/2012 21:04, Mc GRATH Ricardo ha scritto: How about to change tone list on indications.conf file? As I already said indications.conf doesn't work for dahdi channels, unfortunately the callwaiting tone is hardcoded in asterisk itself (not even in dahdi/libtonezone!). I solved patching and recompiling asterisk: http://www.spinics.net/lists/asterisk/msg153399.html Another good solution is to disable call waiting and using local channel instead, as suggested by Warren Selby: http://www.spinics.net/lists/asterisk/msg153424.html Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove the call waiting tone without disabling callwaiting?
Thank you, I already considered such an approach but the customer wanted
to receive the new call *immediately* after the hangup (basically
because it was possible with the old pbx).
This is how I solved: http://www.spinics.net/lists/asterisk/msg153399.html
Such a way I hear the annoying beep every 40 seconds. If someone knows
how to COMPLETELY REMOVE the fucking beep please let me know: there are
already tons of phones ringing everywhere so there is no need for an
annoying beep.
P.S.
By the way, I had a nice time with SNOM phones crashing and freezing
everywhere thanks to a bug while handling SIP ANSWERED ELSEWHERE. I
suggest everyone to upgrade to 8.7.3.15 beta.
Cheers,
Niccolò
Il 02/10/2012 21:16, Warren Selby ha scritto:
Niccolo,
This is what I did for one of my clients. They had a very busy queue,
and were getting annoyed with the Call Waiting beeps. To resolve this,
we changed the method for contacting the agents to Local Channels. The
local channel would then do a check (using the GROUP() function) and see
if it was already in a call or not, and if it was, it would delay
sending the call to that agent. It would then try again after a certain
amount of time had passed.
The agents are added to the queue dynamically using
AddQueueMember(${queue-name},Local/${agent-exten}@agent-callsSIP/${state-exten}).
We would load the appropriate variables in the preceding dialplan.
Here's the snippets from extensions.conf:
[agent-calls]
;Context to dial agents when calls come into their queues
exten = _,1,Wait(1)
exten = _,n,Set(GROUP()=${EXTEN}-calls)
exten = _,n,GotoIf($[${GROUP_COUNT(${EXTEN}-calls)} 1]?wait_longer)
exten = _,n,Dial(SIP/${EXTEN})
exten = _,n,GotoIf(${DIALSTATUS}=UNAVAILABLE?wait_longer)
exten = _,n,Goto(1)
exten = _,n(wait_longer),Wait(15)
exten = _,n,Goto(1)
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[asterisk-users] How to remove the call waiting tone without disabling callwaiting?
Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove the call waiting tone without disabling callwaiting?
Il 01/10/2012 17:12, Danny Nicholas ha scritto: I would start here http://www.voip-info.org/wiki/view/Asterisk+indications+default You could change the tone to something less annoying or just inaudible. Does it affect dahdi channels? If I recall correctly the dahdi tones are hardcoded/placed elsewhere. Thanks, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove the call waiting tone without disabling callwaiting?
Il 01/10/2012 17:47, Danny Nicholas ha scritto: Maybe /etc/asterisk/chan_dahdi.conf No, the only option here is to enable/disable callwaiting. Cheers, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove the call waiting tone without disabling callwaiting?
Is it hardcoded in zonedata.c, am I right? -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove the call waiting tone without disabling callwaiting?
I modified the Italian zone in zonedata.c from
{ DAHDI_TONE_CALLWAIT, 425/400,0/100,425/250,0/100,425/150,0/14000 },
to
{ DAHDI_TONE_CALLWAIT, 0/14 },
but I can still hear the damn beep :(
I even rebooted the pc, suggestions?
Niccolò
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Re: [asterisk-users] How to remove the call waiting tone without disabling callwaiting?
Il 01/10/2012 20:08, Danny Nicholas ha scritto: This is probably a dumb question, but your country/zone is set to it (installs as us by default)? Obviously :) Anyway I think I found where the fucking bastard is hardcoded: chan_dahdi.c in asterisk :) I will change #define CALLWAITING_REPEAT_SAMPLES ((1 * 8) / READ_SIZE) /*! 10,000 ms */ to #define CALLWAITING_REPEAT_SAMPLES ((4 * 8) / READ_SIZE) /*! 40,000 ms */ Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup calls coming from queues
Il 26/04/2012 16:04, Mark Michelson ha scritto: What is the strategy of the queue? Ringall. How are the queue members listed (i.e. are they SIP channels or local channels)? There is only one member listed: SIP/phone-200 My suspicion is that the queue is simultaneously dialing local channels in contexts [context-100] and [context-200]. Since there are no ringing channels in context [from-my-sip-provider] there are no calls to pick up there. However, since [context-100] and [context-200] both have ringing channels, doing a call pickup in either of these results in a successful pickup. I didn't use local channels in the queue members, I listed only one SIP member (SIP/phone-200). That's quite strange, but I have tons of contexts, gotos and include in my real dialplan, way too complex for such tests :( Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Hi, Il 18/04/2012 00:39, Kevin P. Fleming ha scritto: You guys know that it works in Asterisk 10, but you say you can't use Asterisk 10 for some reason that I don't understand. 1) No Debian packages for v10. If you have to maintain lots of servers, installing from sources is a big burden. Compile, install and forget isn't the way I work: if I have to apply a fix or close a security hole I can easily push the patches to my build server which will recompile all the branches I maintain, then every server will automatically upgrade with cron jobs. 2) A new whole of problems when upgrading production machines from a working 1.8.x to v10. That will mean parsing configs manually, find the problems and fixing them. 3) Third parties utilities/hardware/modules. I'm still waiting for a fix for my Sangoma BRI card which did broke when upgrading... You need a compatible version of third parties components to use recent versions of asterisk/dahdi/whatever and upgrading third parties components does always mean problems. 4) Isn't v10 supposed to be beta/non-production/non-long-term-support?[1] If we want to honor what Digium says we should use 1.8 for production servers when reliability is important. Backporting a single unstable feature is much better than the whole thing. 5) What was the purpose of the t38gateway-1.8 branch? Why did it existed at all if not to allow users to use t38 gw in production servers? I even read about the possibility to backport t38 gw to 1.8 as a plugin, but it seems it isn't a requested feature (which is strange because I know peoples who stopped using asterisk because of the lack of t38 gw). I really don't want to do polemics: I always used pstn for the faxes until now and I will keep using it. No problem. Cheers, Niccolò [1]https://wiki.asterisk.org/wiki/display/AST/Proposed+changes+to+Asterisk+release+and+support+cycles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Il 18/04/2012 14:50, Kevin P. Fleming ha scritto: Do you expect Debian-style packages to include these third-party components in Asterisk? If you are talking about DAHDI specifically, moving to Asterisk 10 does not change DAHDI requirements at all. No, I just pointed out that upgrading to a new asterisk version (ie 1.6 - 1.8) can lead to regressions when using third parties components. For example two years ago there was a bug with sangoma cards and asterisk 1.8 and now there is another one with dahdi 2.6. If you feel that having a discussion about what makes sense for users to do and not to do is 'polemics', then fine, you can do whatever you like. Just please stop trying to assign blame or fault to people because this old, unsupported branch doesn't do what you want, especially when there is a current, fully supported release that will do what you want. I think you misunderstood: I just wanted to point out that *I do not blame anyone*, I was just speaking about the reasons because of I prefer to not upgrade to v10. About asterisk 10, it seems I misunderstood the new release cycle, my fault. Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Il 18/04/2012 14:50, Kevin P. Fleming ha scritto: we'll get this corrected That's an awesome news indeed. Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Il 17/04/2012 01:10, Niccolò Belli ha scritto: Tomorrow I will try without directmedia=yes. Unfortunately it didn't help. Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM phones? Please test this patch (broken hints with notifycid=yes)
Patch: https://issues.asterisk.org/jira/secure/attachment/42605/local_remote_hint2.diff https://issues.asterisk.org/jira/browse/ASTERISK-16735?focusedCommentId=191720page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-191720 It's already merged in asterisk 10.4-rc1, it breaks hints for me but I suspect it may be a snom's bug. Cheers, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup calls coming from queues
I suspected it, but it didn't work at first. I fear I didn't understand what the context refers to in Pickup(extension[@context]). I will make an example: phone-100 wants to pick up a ringing phone-200 (call comes from my-sip-provider). This is my sip.conf [phone-100] context=context-100 [phone-200] context=context-200 [my-sip-provider] context=from-my-sip-provider This is my extensions.conf [context-100] exten = test,hint,Queue:MyQueue exten = test,1,Pickup(myphonenumber@from-my-sip-provider) [...] [context-200] [...] [from-my-sip-provider] exten = myphonenumber,1,Queue(MyQueue,r) same = n,Hangup() I expected to use from-my-sip-provider as context in Pickup, unfortunately it didn't work. So I tried both context-100 and context-200 as context in Pickup and they *both* worked! What's the logic behind Pickup's context? Thanks, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Hi, Il 16/04/2012 22:50, Larry Moore ha scritto: Do you have directmedia=no in your SIP configuration? Yes I have. Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup calls coming from queues
Il 20/01/2012 20:32, Alec Davis ha scritto: This maybe not what you want. Our solution was monitor a queue with a BLF, instead of a queue member This reviewhttps://reviewboard.asterisk.org/r/1619/ allows a BLF lamp to flash when a queue is ringing, then the queue can be picked up by the BLF button. The hint does work very well but I really didn't understand how you did pickup the call in example 2... I don't have static members and each (dynamic) member may be logged to another queue too. So even if I know SIP/155 is a (dynamic) member of Queue1 I can't pick up 155 if it's ringing because of another call coming from Queue2. Thanks, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T38 gateway issue
https://issues.asterisk.org/jira/browse/ASTERISK-19684 Can someone help me? It's an old problem I have since the earlier patches which isn't still solved. I'd like to use T.38 gw in production... Cheers, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Hi, If someone is interested I made Debian Squeeze Packages: http://www.linuxsystems.it/2012/04/asterisk-1-8-11-0-debian-squeeze-packages-with-t-38-gateway-queue-hints-and-fixed-rfc4235/ Niccolò Il 30/03/2012 17:22, Niccolò Belli ha scritto: http://www.linuxsystems.it/2012/03/new-t-38-gateway-patch-against-asterisk-1-8-11-0/ I made a new patch from irroot's branch and I ported it to 1.8.11. Unfortunately latest one is still against 1.8.8 and porting from subversion is quite time consuming, hopefully my work will be useful to someone else. Today I had no time to properly test it, so feedbacks are welcome. Squeeze debian packages with t38 gateway will follow. Cheers, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T.38 gateway patch against Asterisk 1.8.11.0
http://www.linuxsystems.it/2012/03/new-t-38-gateway-patch-against-asterisk-1-8-11-0/ I made a new patch from irroot's branch and I ported it to 1.8.11. Unfortunately latest one is still against 1.8.8 and porting from subversion is quite time consuming, hopefully my work will be useful to someone else. Today I had no time to properly test it, so feedbacks are welcome. Squeeze debian packages with t38 gateway will follow. Cheers, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 gateway patch against Asterisk 1.8.11.0
Il 30/03/2012 19:29, Ryan Wagoner ha scritto: It looks like the patch is a backport of the t.38 gateway functionality in Asterisk 1.10. Yes it's a backport from asterisk 10, asterisk 1.8 does not have the t38 gateway functionality and there is no chance to get t38 gw in 1.8 at this point. Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 gateway patch against Asterisk 1.8.11.0
Il 30/03/2012 19:16, Bryant Zimmerman ha scritto: Why is it not in Jarr? Ok I linked it in jira, anyway I don't know how many peoples still follow the old bug report considering it has been closed. Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inclusion of the ILBC codec starting from asterisk 1.8.10/10.2
I noticed it by chance while digging into the code, I think it's a great news! Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cell Phone as a Queue member
Hi, Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries to call SIP/$TRUNK instead. Cheers, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell Phone as a Queue member
Il 31/01/2012 15:42, C F ha scritto: Use local channel Thanks, I completely forget about local channel. Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell Phone as a Queue member
Il 31/01/2012 15:46, Doug Lytle ha scritto: You'll also want to keep track of the number of active calls, since, I believe, the queue app will not be able to see signaling on that line. I'm sorry but, how to? Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proposed changes to Asterisk release and support cycles
I like it! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup calls coming from queues
Il 25/01/2012 22:52, Michael Keuter ha scritto: Outcry! :-) I'm outcrying too :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup calls coming from queues
Il 23/01/2012 21:03, Olivier ha scritto: How can I test this solution on a 1.8.8.1 system ? If I'm not mistaken, diffhttps://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1. Are you sure? Hopefully I will test it in the week end. Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup calls coming from queues
Hi, I have some phones monitoring several extensions, I want them being able to pickup calls using Busy Lamp Field. Unfortunately it doesn't work when the calls come from a queue. Example: Phone 110 wants to monitor phone 102. Phone 102 is a member of the queue Test, it has been added to this queue using AddQueueMember. A call comes from the ISDN and goes to the Test queue, phone 102 starts ringing. Phone 110 sees 102's Busy Lamp Field blinking but can't pick up the call. Please help me, I really need this feature. I'm using asterisk 1.8.7. Thanks, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Login agents on asterisk startup according to hints state
Hi, I did map a key in each phone to add it to the incoming call queue (using AddQueueMember). It also updates a custom hint state for the Busy Lamp Field (BLF). When I restart asterisk it keeps the previous Hint states (I don't know how, but it does), but obviously the phone is no longer a member of the queue. Is there a way to add phones to the incoming queue on asterisk startup according to the hint state? Thanks, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [hint state][BLF] Asterisk 1.8.7 does not send RINGING notifications, even with notifyringing=yes
Hi, I set verbose to 3, but I do not see any RINGING notification in the CLI. On the contrary, when the phone goes UNREACHABLE I get: [Dec 13 21:10:06] NOTICE[9988]: chan_sip.c:25533 sip_poke_noanswer: Peer '152' is now UNREACHABLE! Last qualify: 130 == Extension Changed 152[blf] new state Unavailable for Notify User 154 [Dec 13 21:11:08] NOTICE[9988]: chan_sip.c:20196 handle_response_peerpoke: Peer '152' is now Reachable. (528ms / 2000ms) == Extension Changed 152[blf] new state Idle for Notify User 154 If I manually set the state to RINGING using Set(DEVICE_STATE(SIP/152)=RINGING) I get: Extension Changed 152[blf] new state Ringing for Notify User 154 but otherwise I don't get any ringing notification. Cheers, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 fax detection using g729
Il 19/04/2011 23:41, Kevin P. Fleming ha scritto: If you are the receiver of the call (and thus they are the sender of the call), it is *your* system's responsibility to initiate the switch to T.38, not theirs. Are you sure? So what's faxdetect=t38 for? Cheers, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: No Internet, no asterisk
Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Cheers, Darkbasic P.S. Why nobody ever fixed this annoying bug? Is there a special reason behind? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Internet, no asterisk
Hi, this is an old outstanding problem, unfortunately I don't remember how to walkaround it. I use asterisk 1.8.3 and I have a public IP in my network interface. As soon as the Internet connection goes down, phones stop working. I want to be able to use pstn, isdn and the gsm gateway even if the Internet connection goes down, how can I achieve it? Thank you, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] 802.11x roaming
Hi, I'd like to replace my DECT + fxs phones with some wireless phones. Main problem is: what about the roaming from one ap to another? Do someone adopted the 802.11r standard? What APs and what phones do you suggest me? I'm open to suggestions but I'd like to avoid proprietary solutions, I don't want to be bound to any hardware vendor. My switch does support spanning tree protocol. Thank you, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 fax detection using g729
Il 14/04/2011 12:25, Larry Moore ha scritto: I made a suggestion on how you could check this i.e. have your incoming call go directly to the fax extension, my 1.8.3.2 installation immediately negotiates a T.38 connection in this sceanrio, of course I enabled the fallback option so it will use g711a if T.38 is not accepted by the peer. Yeah, I know, I just didn't have time to check it. Now I've just finished checking and Eutelia DOESN'T send any T.38 re-invite :( I hoped it was an asterisk bug because there was a chance it may be solved, now that I know it's an Eutelia problem I know it will never be fixed. I sent them an e-mail but I already know the answer: T-38-what? -.- Unfortunately it's the only provider which gives unlimited free Italian numbers. Thank you, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 fax detection using g729
Il 14/04/2011 14:34, Larry Moore ha scritto: allow=alaw,g729 ; alaw required for T.30 facsimile if T.38 fails to I didn't understand the point. If you enable both alaw and g729 it will simply use the preferred one: if it's g729 tone based detection will still not work, if it's alaw I will still use that bandwidth expensive codec. Please explain me because I can't get the point. Cheers, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T38 fax detection using g729
Hi, I continue the discussion from https://issues.asterisk.org/view.php?id=19103 If T.38 reinvite detection should still work, why it doesn't? If I use faxdetect = t38 it does never detect the fax, even using alaw. Cheers, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 fax detection using g729
Il 13/04/2011 19:54, Larry Moore ha scritto: That is because the remote endpoint, eutelia, will need to detect the Fax Tones and send the T.38 ReINVITE to you, they may not have T.38 enabled. Uhm... it's very unlikely. As a suggestion you could configure your incoming calls from eutelia to go directly to the fax receive function whilst having the g729 codec enabled, I expect you will then see T.38 re-invite come from Asterisk. See attachment from bug https://issues.asterisk.org/view.php?id=19100 Doesn't Eutelia send a T.38 re-invite there? Anyway I will check again making a packet dump to be sure. What is in your configuration for 159? [159] language=it type=friend qualify=yes ;ping per controllo stato dtmfmode = rfc2833 context=phones-sip host=dynamic disallow=all allow=g729 secret=mysecret Cheers, Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with woomera (ISDN BRI) and playback app: Dropping incompatible voice frame
Hi, when I receive a call from ISDN BRI (with a Sangoma A500) and I try to playback something I get the following error: **[WOOMERA]** HW DTMF supported s1c1- -- Executing [number@from-pstn:1] Answer(WOOMERA/g1/1-7b29, ) in new stack **[WOOMERA]** +++ANSWER WOOMERA/g1/1-7b29 -- Executing [number@from-pstn:2] Playback(WOOMERA/g1/1-7b29, pbx-invalid) in new stack **[WOOMERA]** +++SETOPT WOOMERA/g1/1-7b29 -- WOOMERA/g1/1-7b29 Playing 'pbx-invalid.gsm' (language 'it') [Apr 6 19:12:18] NOTICE[1434]: channel.c:4046 __ast_read: Dropping incompatible voice frame on WOOMERA/g1/1-7b29 of format alaw since our native format has changed to 0x2 (gsm) I don't have this problem when I receive calls from PSTN (Sangoma A200). Can someone help me? Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!
What about transparent t.38 gatewaying? :-( -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
I do not use windows on the desktop/laptop, but when I have to I use putty. Darkbasic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday at 1PM: SIPVicious has a new tool: svcrash
Awesome! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to limit the calls leaving a queue?
No one can help me? Darkbasic Il 18 ottobre 2009 15.49, Niccolò Belli darkbas...@gmail.com ha scritto: 2009/10/17 Paul Hales pdha...@optusnet.com.au: I have used the group function to limit the calls entering a queue for a similar reason to yourself. But I do not want to limit the calls entering a queue (I can already do it with maxlen= in queues.conf), people should wait in the queue until an operator hang up. Is it possible to do it? Darkbasic ___ -- 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] how to limit the calls leaving a queue?
Hi, I already considered such a solution: I will have lots of loops and the customer will loose its position if someone else who called later will find the queue 'ready to join' before him. I will also have a music on hold mismatch when he will join the queue and I will loose all the benefits of the asterisk queue management! If someone else will confirm me there is no other way to do it maybe I should file a wishlist bug in the asterisk tracker, beacuse I'm sure I'm not the only one who need such a feature. Cheers, Darkbasic 2009/10/20 Miguel Molina mmol...@millenium.com.co: With the standard options your are right, you can limit the incoming calls that enter the queue. But AFAIK, there's no way to limit how many calls leave the queue. You only have a timeout to control when a call leaves the queue, not how many of them. With that been said, I only can think of a dialplan solution, that could fit your needs or not: 1. Queue your call normally, with the standard timeout you want 2. If the call leaves the Queue, it will continue dialplan execution at next priority, and there you can make it be part of a group of channels. 3. Count that group and check if it reached the limit you need. If above the limit, requeue your call to the same queue it was. That's all I can think of, without a hard work of messing with app_queue.c source code. Hope you get the idea. ___ -- 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] how to limit the calls leaving a queue?
2009/10/17 Paul Hales pdha...@optusnet.com.au: I have used the group function to limit the calls entering a queue for a similar reason to yourself. But I do not want to limit the calls entering a queue (I can already do it with maxlen= in queues.conf), people should wait in the queue until an operator hang up. Is it possible to do it? Darkbasic ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] how to limit the calls leaving a queue?
Hi, I explain what I want to do.. All the operators share their phones. The number of the operator isn't constant, so it's possible that two operators share all the phones. They need to move all around, so they pick up the first phone they find. If there are only few operator is very annoying for them to ear the other phones ringing while they are at the phone! So I'dd like to limit the maximum number of simultaneous calls leaving the queue, but how to do it? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
