[Asterisk-Users] Agent penalties and busy status
Hi all, When implementing a queue using members like this: [queue] strategy = rrmemory member = Agent/1000,1 member = Agent/1001,1 member = Agent/1002,2 member = Agent/1003,2 member = Agent/1004,2 And you call into the queue, agents 1000 and 1001 will ring in an alternating fashion until one of them answers it. You might have seen my question coming already, so I won't delay it anymore: is it possible to have 1000 and 1001 only ring once and then fallback to the other penalty-levels? With disciplined agents it's no problem, but when 1000 and 1001 decide to not answer any calls for a while (circuit-busy / noanswer), the rrmemory strategy doesn't fail over to other agents and the whole queue is stuck. If it isn't possible, I would be happy to change this in the Asterisk code for my site installation, but where should I start hacking? Any pointers are greatly appreciated. Thanks in advance for your time. With kind regards, Frank Schoep ___ 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] Unattended Agent Login
Hello all, I've got a question regarding automatically logging in an agent on a predefined extension. I want to provide agents with a webinterface they can use to login in, now you might wonder why they can't use the phone - don't ask - customers eh :-) So what I basically need is a way to log in an agent using AgentCallbackLogin at an extension without them having to answer / pickup a phone to do so. I looked at the Manager API but did not find any command related to agent logins. I then thought up the possibility to login agents using predefined extensions, to which I will add the AgentCallbackLogin command and later use Originate using the Manager API to set up the calls. Has anyone got experience doing the agent login in that way or some tips on how to do the actual implementation? Thanks in advance for your time. Sincerely, Frank ___ 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] Sharing variables between contexts
Hello all,
I'm having trouble getting variables to work the way I want them to, let me
begin with a simple explanation of the problem, I'm using something like this
in my extensions.conf:
[default]
exten = 5000,1,SetVar([EMAIL PROTECTED])
exten = 5000,2,Goto(mailexten,s,1)
exten = 6000,1,SetVar([EMAIL PROTECTED])
exten = 6000,2,Goto(mailexten,s,1)
[mailexten]
exten = s,1,System(/mail.sh ${Recipient})
exten = s,2,Hangup
As an unsuspecting user, I thought this would work - the variable Recipient
should be available in the [mailexten] context, but apparently this is not
the case. I'm using Asterisk 1.0.9, is this a known problem or am I just
expecting the wrong thing?
Sincerely,
Frank
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Re: [Asterisk-Users] Sharing variables between contexts
On Monday 11 July 2005 14:33, jurczak wrote:
I am having the same thing on my extensions.conf and it works fine. I am
using Asterisk 1.0.7
Is it possible that using queues causes problems with regard to handling
variables? It seems that variable handling between contexts is broken after
an incoming call has gone through a queue. Even handling variables within a
context seemed to go foobar after a queue, at least it did when I tested it.
I'll try investigating what causes the problem, but I've currently implemented
a workaround using the ${CALLERIDNAME} channel variable.
Sincerely,
Frank
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Re: [SPAM:***** SpamScore] RE: [Asterisk-Users] Call Transfer using SIP clients
On Tuesday 05 July 2005 05:57, Tulika Pradhan wrote:
call transfer works for me fine without any additions in features.conf
by simply using Dial(SIP/${EXTEN},20,tT)
and pressing #number to be transfered to
this works both from caller as well as callee.
tulika
Could you provide me with some more information so I can check where the
differences in our setups are? It would really help to see how you
implemented your extensions and SIP configuration. Could you describe the
following regarding your Asterisk installation:
- Asterisk version
- The SIP clients you use
- Excerpt of extensions.conf, which definitions and contexts do you include
- Excerpt of features.conf, which lines (if any) are in there
- (Maybe) an excerpt of sip.conf, how are the SIP peers configured
I hope you find the time to post these bits and pieces as it will make it
easier for me to debug the situation. I've already tried numerous settings
and combinations of options, but haven't had any luck yet. Thanks in advance
for your precious time.
If anyone else has some ideas regarding my question, feel free to jump in -
the more the merrier.
Sincerely,
Frank
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Re: [SPAM:***** SpamScore] Re: [Asterisk-Users] Call Transfer using SIP clients
On Monday 04 July 2005 16:47, Elwin Andriol wrote: I don't know if this will be of any help to you, but at least I can confirm problems with transfering calls with SIP agents. A little while ago we were having big problems getting transfers using DTMF to work. In that particular situation we were using a mix of only hard SIP devices (BT101's and Elmeg/Snom190's) and ZAP channels. We tried both the stable version of asterisk and the CVS HEAD, but without results (but negative). In the end, we solved the problem by not using DTMF transfers at all, but by using the transfer capabilities of the SIP devices themselves (transfer for and hold buttons). These buttons did not appear to work (correctly) with the stable asterisk version we initially used (1.0.7), but with the CVS HEAD ( 29-MAY-2005) they appear to work just fine. I'm not familiar with soft SIP agents, so I don't know if the ones you use have such build-in transfer capabilities as their hardware counterparts like the BT101's and Snom190's have. I they do, you might wan't to give it a try. This is of course rather a workaround than a solution to your problem. E. Andriol The X-Lite softphone does indeed have a Transfer and Hold in the interface, but the functionality of those buttons appears to have been disabled when the client is connected and registered on the Asterisk server. Pressing the on-screen buttons doesn't have any effect while either having a call or while idle. Related to client-side transferring: I set the canreinvite option in the SIP configuration to no because both clients are behind a NAT / firewall and I read in the documentation that you'd want to disable the canreinvite option in those situations. I haven't had any trouble because of this, as I stated earlier calling and talking is working without hitches. I haven't had the chance to try hardware phones yet, the testing I'm doing at the moment involves softphones only. Now that I think of it, I'll try to setup other applications again which might send DTMFs in a different form compared to X-Lite. In the meantime thanks in advance to everyone involved in this thread now and in the future. Sincerely, Frank ___ 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: [SPAM:******** SpamScore] [Asterisk-Users] Call Transfer using SIP clients
On Monday 04 July 2005 16:11, Frank Schoep wrote: I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. ... The SIP setup is working without a problem, the X-Lite application correctly registers the users and I can set up calls between them. I've also tested queues and they work without a problem, too. ... I searched the web and the mailing list archive for a solution, and if I recall correctly, someone stated that call transfer is only available for calls originating from the PSTN. Is this correct, also in regard of the current version of Asterisk? Has anyone got an idea how to get call transfer to work? ... One thing I tried was to change the DTMF settings in the clients, so they are sent in-band, but this also didn't help. OK, I'm happy to report that I finally managed to get the call transfer to work with a different softphone application called LinPhone. I will investigate why X-Lite doesn't work with sending DTMFs, but at least I know now that my Asterisk setup isn't at fault. Sadly, I can not share a solution any more detailed than this, the softphone switch was enough to convince me that the Asterisk server is working perfectly, it was primarily my X-Lite configuration at fault, so my sincere apologies for posting this on the Asterisk mailing list. If I find out how to get it working, I will append that information to the thread so others can reuse that knowledge later on, I'm sure someone will appreciate it. Thanks to each and all contributors who've got me thinking about finding the problematic part in my setup. Sincerely, Frank ___ 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: [SPAM:***** SpamScore] Re: [SPAM:******** SpamScore] [Asterisk-Users] Call Transfer using SIP clients
On Tuesday 05 July 2005 09:29, Frank Schoep wrote: If I find out how to get it working, I will append that information to the thread so others can reuse that knowledge later on, I'm sure someone will appreciate it. So, I just got X-Lite working alongside Asterisk, the problem was (call it a premonition) the fact that I set them up to send DTMFs in band. Setting this option to disabled made the X-Lite softphone work flawlessly. I hope that helps someone. Sincerely, Frank ___ 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] Call Transfer using SIP clients
Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently using Asterisk CVS as of July 4th 2005. The only means of communication at the moment is the XTen X-Lite SIP Client, I already added the following entries to my sip.conf configuration file: [frank] canreinvite=no type=friend secret=frank username=frank nat=yes host=dynamic [test] canreinvite=no type=friend secret=test username=test nat=yes host=dynamic The SIP setup is working without a problem, the X-Lite application correctly registers the users and I can set up calls between them. I've also tested queues and they work without a problem, too. Next up is my extensions configuration, of which the interesting section now looks like this: [default] include = general ; longshot, added out of desparation include = parkedcalls ; longshot, added out of desparation include = featuremap ; longshot, added out of desparation exten = 800,1,Answer exten = 800,2,Dial(SIP/frank,20,tT) exten = 800,3 Hangup exten = 802,1,Answer exten = 802,2,Dial(SIP/test,20,tT) exten = 802,3 Hangup Notice the inclusion of several contexts that should or would have to be defined in the features configuration. My features.conf looks something like this, I trimmed the 'general' section for brevity: [general] ; (trimmed) default options [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer My testing scenario starts as follows: - log in both X-Lite SIP clients - from the 'test' phone, call extension 800 - on X-Lite client 'frank' accept the call - talk to eachother At this point I want to transfer to call to another extension, also defined in sip.conf but unlisted here. The problem is that nothing happens when I press the #1 or *2 keys in the 'frank' X-Lite client. I also tested these key combinations on the 'test' X-Lite client during the call, but that also had not effect. I searched the web and the mailing list archive for a solution, and if I recall correctly, someone stated that call transfer is only available for calls originating from the PSTN. Is this correct, also in regard of the current version of Asterisk? Has anyone got an idea how to get call transfer to work? One thing I tried was to change the DTMF settings in the clients, so they are sent in-band, but this also didn't help. Should I revert this option? Thanks in advance for your time and patience. Sincerely, Frank Schoep ___ 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: [SPAM:***** SpamScore] [Asterisk-Users] Join wav Files in Linux
On Monday 04 July 2005 16:31, Kevin Kiely wrote: Does anyone know how to join two .wav audio files via the command line in Linux for playback with Asterisk? Kevin, you might want to try Sox, see http://sox.sf.net for more information. I'm not sure it can join or concatenate audio files, but I think it will. Sincerely, Frank ___ 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
