Re: [Asterisk-Users] Phone Recommendation.
On Tue, 26 Apr 2005, Dana Olson wrote: You mean like the problem I described earlier on this list? http://lists.digium.com/pipermail/asterisk-users/2005-April/103153.html I am not sure why I didn't think of disabling call waiting, but that seemed to work with a Grandstream BudgeTone phone... I'm doing more testing now. Sounds exactly like the same problem. Of course, the $65 grandstreams allow you to disable call waiting.. The stupid $130+ Polycom's don't. :( -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Phone Recommendation.
On Wed, 27 Apr 2005, Wiley Siler wrote: Actually, I was recently told that the callwaiting disable feature in SIP 1.4.1 was not working. Are you using 1.4.1? If your method works, it would be useful to the other Polycom users I am sure. I love the conf scripts for Polycoms but it is fun chasing down fields in the XML sometimes. I'm using a combination of 1.3.1 and 1.3.4 phones. I looked at the Admin guide for 1.4.1, and like the others, found no mention of a call waiting disable. Nor could Polycom tell me how to disable it. --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Phone Recommendation.
On Mon, 25 Apr 2005, Wiley Siler wrote: Call waiting can be disabled in Asterisk via *71 regardless of the phone used. Cheers, Wiley Well, this is part of a larger problem I'm having. I can't get CheckGroup/SetGroup to work as I think it should for my dynamically added ACD agents. The management here is frustrated, and they just want to buy a few phones that simply can have call waiting disabled. --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phone Recommendation.
I'm looking for recommendations for a office phone that has the ability to disable call-waiting. Needs to be similar in features to a Polycom IP300. Thanks, --Sean -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: CVS-HEAD and CheckGroup/SetGroup
On Wed, 20 Apr 2005, Noah Miller wrote: Is this to disable the call waiting on Polycom phones? If so, I don't think there wouldn't be a need to have it on the s extension. You'd just need to have it on any extension that the Polycom phone would call (other handsets, outside lines, voicemailmain, etc). CheckGroup does increment n+101, but unless you want them to get a busy signal, I'd probably move them on to the next line appearance on the phone, or to a voicemail box. Sending this reply again, because I'm not sure it made it to the list... I never got a copy... I'm only using it to disable call waiting for ACD groups. It works under stable / v1-0 / whatever.. kinda. call waiting is effectively disabled, only for call queues (ACD).. Which is what I want. However, here's the problem that led me to try HEAD. Under stable, When someone in the ACD group takes a call (and the SetGroup is set) and then transfer the call to someone else.. Until the original call is disconnected, that ACD agent will not get any new calls. IE: the SetGroup doesn't go away until the call ends. If someone in my tier 1 department transfers a call to tier 2.. and that call runs for an hour, then I've got a tier 1 tech taking no calls for an hour. This doesn't seem to be a problem when I use head, but then again.. the call waiting problem comes back. I'm on the verge of doing something stupid.. like... just going out and buying a bunch of phones that I can disable call waiting on. Sad, but true.. Anyone want a bunch of very lightly used Polycom phones? --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: CVS-HEAD and CheckGroup/SetGroup
On Wed, 20 Apr 2005, Noah Miller wrote: Is this to disable the call waiting on Polycom phones? If so, I don't think there wouldn't be a need to have it on the s extension. You'd just need to have it on any extension that the Polycom phone would call (other handsets, outside lines, voicemailmain, etc). CheckGroup does increment n+101, but unless you want them to get a busy signal, I'd probably move them on to the next line appearance on the phone, or to a voicemail box. I'm only using it to disable call waiting for ACD groups. It works under stable / v1-0 / whatever.. kinda. call waiting is effectively disabled, only for call queues (ACD).. Which is what I want. However, here's the problem that led me to try HEAD. Under stable, When someone in the ACD group takes a call (and the SetGroup is set) and then transfer the call to someone else.. Until the original call is disconnected, that ACD agent will not get any new calls. IE: the SetGroup doesn't go away until the call ends. If someone in my tier 1 department transfers a call to tier 2.. and that call runs for an hour, then I've got a tier 1 tech taking no calls for an hour. This doesn't seem to be a problem when I use head, but then again.. the call waiting problem comes back. I'm on the verge of doing something stupid.. like... just going out and buying a bunch of phones that I can disable call waiting on. Sad, but true.. Anyone want a bunch of very lightly used Polycom phones? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS-HEAD and CheckGroup/SetGroup
Do the SetGroup and CheckGroup functions behavior differently in CVS-HEAD
vs CVS v1-0?
When I upgrade to CVS-HEAD my call waiting disable doesn't seem to work,
using:
exten = s,1,SetGroup(SIP${ARG1})
exten = s,2,CheckGroup(1)
exten = s,3,Dial(Sip/${ARG1},15,t)
Anyone know why this is?
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
sean a. newton [EMAIL PROTECTED]
louisville, ky, usa http://wewt.net
Another day, another convertible and another hotel
full of cops.-- Hunter S. Thompson
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
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Re: Fwd: [Asterisk-Users] CheckGroup and transfers
On Thu, 31 Mar 2005, C F wrote: -- Forwarded message -- From: C F [EMAIL PROTECTED] Date: Thu, 31 Mar 2005 20:05:39 -0500 Subject: Re: [Asterisk-Users] CheckGroup and transfers To: Sean A. Newton [EMAIL PROTECTED] I don't think it is part of stable, if you are running HEAD make sure it's after 1/4/05. If it is, then report it. I tried applying the patch to a v1-0 CVS release, it errored out, could only apply 2 of 3 hunks. :( I can't get CVS HEAD to build. I tried Friday (4/1), Monday (4/4) and today (4/5). Fun! --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] compilation of asterisk
On Mon, 4 Apr 2005, Alex wrote: Hi guys Trying to compile asterisk and i am receiving this errror. gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 any help will be appreciated. thanks Google is your friend. Install the SSL dev libraries.. libssl-dev --Sean -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS-HEAD compile? Was: Checkgroup and transfers
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue.conf config
On Fri, 1 Apr 2005, Obihuan wrote: Hello all, There are any way for the queue agents in asterisk that they do not need to login in the queue to begin recibing calls? I want to use this queue for our recepcionist, with only one agent. All that I want is, 1. The recepcionist do not need to make a login in the queue. 2. The recepcionist not have to hear the phone all the time waiting for new calls, when she hangs up the phone asterisk make a logout for the agent and she must to login it again to recibe new calls. Use static agents, defined in queues.conf.. Example: [office] strategy = ringall timeout = 600 retry = 5 music = default member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CheckGroup and transfers
On Wed, 30 Mar 2005, C F wrote: I think this bug is what you describe: http://bugs.digium.com/bug_view_page.php?bug_id=0003067 Hope this helps. I think so, but if I'm reading this correctly, the patch is already part of the CVS version? --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email forum??
On Thu, 31 Mar 2005, Chuck Bunn wrote: Hi, I am new to Asterisk and the first thing I have noticed about Asterisk and Pingtels open PBX's is that they are using this dinosaur method of running forums. It is a real pain getting every message in the forum and essentially keeping my own database of issues. With that said are there any forums that are well used or that might even convert this email in a true forum that is searchable and that doesn't require me downloading every email. Before you go and rant on me go see how Mambo Server does it at http://forum.mamboserver.com. The forums are easy to use and thus are easy to participate in. I use mozilla Thunderbird and I have setup filters and all but it still is a pain to use this outdated email forum. Thanks I've actually been meaning to setup a forum... just haven't had time. I already registered the domain (asteriskusers.com), hopefully soon I'll have some free time to setup the DNS and install vBulletin. --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CheckGroup and transfers
Hello, I'm using SetGroup and CheckGroup to work around the Polycom's call waiting issues and calls coming into a queue.. I've found a problem, and I can't think of a way to fix it. When someone calls into a queue, and my agent transfers the call to another person in the company, that agent does not get any more calls in the queue. Once the call terminates and the channel data is destroyed, the agent can get calls again. Anyone run into this? --Sean -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wrapuptime + agents.conf
voip technocrat [EMAIL PROTECTED] wrote: my aim is every call needs have wrapuptime of 5000 ms but when ever a call comes its directly connecting not wating any more. your views will be highly regarded with regards I'm using AddQueueMember.. for me, wrapuptime only seems to work from the beginning of the call, not the end. However, I'm defining wrapuptime in queues.conf, not agents.conf, since my agents are dynamic. I know in agents.conf it says that wrapuptime is in MS, but from what I can tell, if you define it in queues.conf, it's seconds. If the call exceeds the length of the wrapuptime value, there is no wrapuptime. So if you set it to 120, and the call only lasts a minute, the next caller will wait another minute before being connected. Likewise, if you set it to 120, and the call runs 135 seconds, there's no wrapuptime. I'd say that's probably a bug.. At this point it's strictly an annoyance, but I'd love to hear suggestions from the list. --Sean -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re-write callerid?
On Fri, 25 Mar 2005, Remco Barende wrote:
Is it possible to rewrite caller id's?
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but surely there
must be a better solution?
Thanks!!
Remco
I set the Caller*ID before I place the outgoing call, like so:
exten = _91NXXNXX,1,SetCallerID(IgLou Internet 5029663848)
exten = _91NXXNXX,2,Dial(Zap/r1/${EXTEN:1})
Hope that helps,
--Sean
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
sean a. newton [EMAIL PROTECTED]
louisville, ky, usa http://wewt.net
Another day, another convertible and another hotel
full of cops.-- Hunter S. Thompson
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
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Re: [Asterisk-Users] Re-write callerid?
On Fri, 25 Mar 2005, Remco Barende wrote:
exten = _91NXXNXX,1,SetCallerID(IgLou Internet 5029663848)
exten = _91NXXNXX,2,Dial(Zap/r1/${EXTEN:1})
Yes, but this way you can only display one single phone number, and not
the MSN number for each SIP phone?
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the company
but 0031201208 to the outside.
I see.. I can't think of an easy way to do that, short of an AGI script
that checked a flat file or database. You'd pass the local extension to
the AGI and it'd return the MSN number.
--Sean
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
sean a. newton [EMAIL PROTECTED]
louisville, ky, usa http://wewt.net
Another day, another convertible and another hotel
full of cops.-- Hunter S. Thompson
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Re: [Asterisk-Users] Two companies - One Asterisk???
On Fri, 25 Mar 2005, Noah Silverman wrote:
We have two small business that run out of our office. One business has
3 phone lines, and the other has only one.
In a perfect world, Asterisk would indicate WHICH line (or group) the
outsider caller called, so that we would know which way to answer the
phone. The incoming calls would go through a voice mail menu first -
different for each company. (Press one for sales, two for accounting,
etc.) Whatever they pressed, the correct extension would ring, show the
caller id correctly, AND indicate which outside line (or group) the call
came in on.
I'm running the Polycom IP500 phones. I think, from Noah Miller's
suggestion, that there might be a way to use the line buttons on the
phone. Tie a group of lines to button one, and a group of lines to
button two. I just don't know HOW to do this. (For the record, I'm
using the Zap interface for incoming POTS lines.)
I'm running two companies on one * box, however I'm using PRIs, so it's
easy to break out everything.
We have seperate menus, based on what number you dial, and all that jazz.
One idea if you're sending calls from both companies to one person, you
might prepend something onto the CallerIDName..
For example, I have techs that mainly answer technical support calls, but
if sales or billing is not available, they end up with the call.
As such, Tech support calls are tagged with
SetCIDName(TECH:${CALLERIDNAME}) , same with sales, and billing..
Some of us also have DID numbers that ring directly to our desks, I tag
these with DID:${CALLERIDNAME}
Hope that helps...
--Sean
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
sean a. newton [EMAIL PROTECTED]
louisville, ky, usa http://wewt.net
Another day, another convertible and another hotel
full of cops.-- Hunter S. Thompson
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[Asterisk-Users] IP300 soft key configuration
I'm trying to reconfigure my IP300 softkeys.. Currently when on a call, I have to hit more and then transfer.. I'd like make transfer appear on the first screen. Right now there's hold on there.. and hold is kind of redundant, since the IP300 has a hard hold button. I tried doing it in the keys/ section of ipmid.cfg, but it doesn't seem to work.. anyone done this or something similar? I checked the admin guide but am confused.. I think I'm supposed to be using the key.x.y.function.prim thing, but it doesn't work for me.. Thanks in adavance.. --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom and call waiting again..
On Thu, 27 Jan 2005, Kevin P. Fleming wrote:
Adam Goryachev wrote:
[local-stuff]
; This is where we pretend a channel is an extension
exten = 1234,1,SetGroup(SIP1234)
exten = 1234,2,CheckGroup(1)
exten = 1234,3,Dial(SIP/1234,15)
exten = 1234,104,Busy
[queue-stuff]
exten = 6939,1,AddQueueMember(Local/${CALLERIDNUM})
You are close... that should be:
AddQueueMember(Local/[EMAIL PROTECTED])
That way when the queue app tries to call the agent, it will have an
extension _and_ a context to deliver the call to.
Excellent, I will look into this. :D
Thanks Guys!
--Sean
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Re: [Asterisk-Users] Re: Polycom and call waiting again..
On Wed, 26 Jan 2005, Noah Miller wrote: Have you tried adding SetGroup(), and CheckGroup() functions to the dialplan that rings the phone? It maybe something to try. I think the problem is that these functions only work from the dialplan. In this case, Sean is trying to get calls from a Queue (and not the dialplan) to the correct line on the phone. I was thinking about implementing a queue for our receptionists, but this problem prevents me from doing that, and I haven't figured out any way around it. Maybe the new 1.4.1 firmware provides a way to disable that horrid call-waiting feature? Has anybody gotten it to run successfully? Exactly.. SetGroup was suggested by someone on the irc channel.. I looked at it briefly. I was then shot down by someone saying to save my effort, it didn't work. I suspected as much, due to the fact that the Queue function doesn't use the exten config for that phone. And it shouldn't.. The phone should be able to take care of this problem.. I've unfortunately got myself into a bind because I've bought ~35 of these phones. :eek: If everyone thinks SetGroup and CheckGroup will work, I will spend the next days working with it, but I don't want to go barking up the tree of something that doesn't look like it will work. :| I'm also interested to try out the 1.4.1 firmware. Just need to procure a copy of it.. --Sean -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom and call waiting again..
On Wed, 26 Jan 2005, Kevin P. Fleming wrote: But you _can_ use SetGroup/CheckGroup/GetGroupCount if you don't put the SIP peer directly into the queue, but instead add a Local/.. channel that makes the Queue call out to the agent via a special context in your dialplan. This special context can then do anything it wants, including returning Busy/Congestion back to the Queue app if needed. I understand the concept of what your saying, but I can't seem to visualize how to implement it. Do you have an example of this? I would very much appreciate it. --Sean -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom and call waiting again..
I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk sends the call to the one agent logged in. The agent answers and is talking to the caller. While on this call, another person comes into the queue. Asterisk immediately tries to send the call out to the Polycom. Of course, since the Polycom has multiple call appearances (call waiting), the Polycom doesn't send back a busy, so the call sits on the phone in call waiting. Meanwhile the agent is getting blinks and beeps from the phone. After the timeout specified in queues.conf expires, it dials the agent again.. I just need some suggestions on what to try. I don't mind doing the research, but I'm out of ideas on what to search for.. Thanks in advance... --Sean -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AddQueueMember and call distribution
Is there anyway to get Agents (members of the queue) logged out if they do not answer a call. I'm assuming that autologoff doesn't work with AddQueueMember. I've searched the wiki and google to death... Currently, when a call comes in to the queue, one member just rings and rings and rings. It never rolls to another member, and the offending member (not answering) never gets logged off, so the call just gets stuck until an ultimate timeout. Thanks for the help... Here's what i've got setup right now, on my test bed. queues.conf [support] strategy = leastrecent timeout=25 retry=5 wrapuptime=15 music = default announce = queue-support agents.conf [agents] ackcall=no autologoff=15 extensions.conf ; Tech Support Queue exten = 1100,1,Queue(support) exten = 1100,2,Hangup ;tech login exten = 81,1,AddQueueMember(support) exten = 81,2,Playback(agent-loginok) exten = 81,3,Hangup ;tech logoff exten = 82,1,RemoveQueueMember(support) exten = 82,2,Playback(agent-loggedoff) exten = 82,3,Hangup -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- sean a. newton [EMAIL PROTECTED] louisville, ky, usa http://wewt.net Another day, another convertible and another hotel full of cops.-- Hunter S. Thompson -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
