Re: [Asterisk-Users] USB handset options for softphones

[Steve Feinstein]

And the opposite is also true.  I've tried a few Skype Certified 
devices, and generally the audio works (though in some cases it's (IMO) 
unnecessarily turned off unless Skype is in a call, perhaps so you don't 
hear the beeps and squeaks a PC normally makes)  And almost universally, 
a Skype keypad can't be used by non Skype clients. 

We were thinking of writing a Skype emulation layer to enable all these 
devices, has anyone already started down this path?


-Steve Feinstein


Martin Joseph wrote:


On Jun 21, 2006, at 10:35 AM, Michael Graves wrote:

With few exceptions a USB phone is just and audio device to the host 
PC. Most will work with any soft phone. The Phoenix Duet and MV900 
that I have used both worked equally well with

Asterisk, Sip Phone/Gizmo, FWD, Firefly and Skype.

There are some Skype Certified hardware devices appearing. These 
usually have buttons that mimic the software buttons on the Skype 
client. In doing that they use the Skype API directly.
They may work as normal handsets with other soft phones, but I've 
never tried this.



The above is sort of misleading information in my opinion.

It's true that any of the USB phones can function as audio devices, 
but if you expect the KEYPAD on the device to work (notice the devices 
he mentions have none), the softphone needs to support that specifically.


Marty

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Re: [Asterisk-Users] Multiple announcements in a queue ??

[Steve Feinstein]

I was thinking of a script that just copied a 
different/random/sequential audio file to it's stdout everytime it was 
called.  Name the pipe with a .gsm (or whatever format you use) 
extension.  And tell asterisk to use the pipe instead of a static file.  
I think in theory it should work.  I'm not sure if there'll be blocking 
issues should the script fail.  But it's something to think about.


-Steve


A.J. Paxson wrote:

On 5/15/06 10:24 PM, Steve Feinstein  wisely said:

  

I haven't thought it through really, so if  it's a bad idea please let
me know.  But I think a named pipe would be a good unix-y way to do this.

-Steve Feinstein

A.J. Paxson wrote:


Hi All!

I've really been struggling trying to get around this.  Instead of the same
announcement being played over and over again, I want to be able to play
more than 1 announcement in a queue.

Does anyone have any brainstorming ideas on how I can try this?

Once a caller is in a queue, I no longer have any control inside that queue.
I can have that queue timeout, play a different announcement, and place them
back in the queue, but then the caller loses it's place.

Any ideas?

~~Aaron

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Thanks for the reply, Steve!

How so?  I understand what you mean by named pipes, but not sure how to use
it in this context.  Care to give a pseudo example of what you are thinking?

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Re: [Asterisk-Users] Multiple announcements in a queue ??

[Steve Feinstein]

I haven't thought it through really, so if  it's a bad idea please let 
me know.  But I think a named pipe would be a good unix-y way to do this. 


-Steve Feinstein

A.J. Paxson wrote:

Hi All!

I've really been struggling trying to get around this.  Instead of the same
announcement being played over and over again, I want to be able to play
more than 1 announcement in a queue.

Does anyone have any brainstorming ideas on how I can try this?

Once a caller is in a queue, I no longer have any control inside that queue.
I can have that queue timeout, play a different announcement, and place them
back in the queue, but then the caller loses it's place.

Any ideas?

~~Aaron

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Re: [Asterisk-Users] USB conference phone

[Steve Feinstein]

Once you plug it in, the computer sees it as a speaker, and mic. So if
you can get OSS or ALSA to use it, you can use it just as you'd use any
sound card in asterisk. 



Kerry Garrison wrote:

  I use a softphone 

  
  
-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of 
John covici
Sent: Thursday, April 27, 2006 9:11 PM
To: Steve Feinstein
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] USB conference phone

OK, assuming the usbaudio sees the conference phone and can 
work it, how would you write an extension to ring that?

on Thursday 04/27/2006 Steve Feinstein([EMAIL PROTECTED]) 
wrote   It's a standard USB audio device.  While I haven't 
tried it, I'm pretty   sure the Linux USB audio driver will 
probably see it.
 
  -Steve
 
  John covici wrote:
   Any way to use this on a Linux box so I could use this 
with asterisk?
   I have a windows box on the same network, but how would 
I get asteriskto see such a thing?
  
   Thanks.
  
   on Wednesday 04/26/2006 Steve 
Feinstein([EMAIL PROTECTED]) wrote  
http://www.iogear.com/main.php?loc=productItem=GPH100U

 I've got a couple of these, they're $40 and there's a 
$20 rebate going  on now. For that price it's pretty 
amazing. Plug and play, no drivers  required. Quality 
very good, it does echo cancellation and noise  
reduction. I only wish it had a mute button.

 BTW: I don't have any affiliation with ioGear other 
than I like this  product.


 Jim Houser wrote:
  Personal preference. I'm not a big headset guy.
  The real point of my reply was to say how impressed 
I am with USB talk   quality when compared to a 
hardphone on Asterisk or our Avaya   Communications 
Manager. Like my wife says, I guess I'm not being   
clear... :) 
--
--
  *From:* [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED]] 
*On Behalf Of *Dean   Collins   *Sent:* 
Wednesday, April 26, 2006 10:24 AM   *To:* Asterisk 
Users Mailing List - Non-Commercial Discussion   
*Subject:* RE: [Asterisk-Users] USB conference phone  


Kerry, do you actually own one? Have you used it 
  

for long? What are   you using it for?
 
  (jim ? personally I cant see the point of using 
your phone when I have   a very good quality headset 
and mic.).
 
  Dean
 
  
--
--
 
  *From:* [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED]] 
*On Behalf Of *Kerry   Garrison   *Sent:* 
Wednesday, 26 April 2006 10:36 AM   *To:* 'Asterisk 
Users Mailing List - Non-Commercial Discussion'
  *Subject:* RE: [Asterisk-Users] USB conference 
phone This is an excellent USB 
speakerphone 
http://voipspeak.net/index.php?option=com_contenttask=viewid

  
  =39Itemid=27
  
  
  
http://voipspeak.net/index.php?option=com_contenttask=viewi

  
  d=39Itemid=27
  
  
 
  
--
--
 
  *From:* [EMAIL PROTECTED]
  
[mailto:[EMAIL PROTECTED]] *On Behalf Of
  *Jim Houser
  *Sent:* Wednesday, April 26, 2006 6:26 AM
  *To:* 'Asterisk Users Mailing List - 
Non-Commercial Discussion'
  *Subject:* RE: [Asterisk-Users] USB conference phone
 
  I don't know about this phone but I can tell 
you I have a vendor
  that will only talk to me via Skype so I purchased this:
  
http://www.provantage.com/usb-internet-phone~220150620.htm
  
http://www.provantage.com/usb-internet-phone%7E220150620.htm
 
  It operates nice and has very good call quality.
 
  
--
--
 
  *From:* [EMAIL PROTECTED]
  
[mailto:[EMAIL PROTECTED]] *On Behalf Of
  *Dean Collins
  *Sent:* Tuesday, April 25, 2006 8:22 PM
  *To:* Asterisk Users Mailing List - 
Non-Commercial Discussion
  *Subject:* [Asterisk-Users] USB conference phone
 
  Has anyone actually used these USB speakerphones
 
  
http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-
free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDV
WQQrdZ1QQcmdZViewItem
 
  Seems to get a pretty good review here
 
  
http://voipspeak.net/index.php?option=com_contenttask=viewid

  
  =39Itemid=27
  
  
  
http://voipspeak.net/index.php?option=com_contenttask=viewi

  
  d=39Itemid=27
  
  
 
  But looking for real world feedback.
 
  Cheers,
 
  Dean
 
  This e-mail and any attachments may contain 
confidential

Re: [Asterisk-Users] USB conference phone

[Steve Feinstein]

It's a standard USB audio device.  While I haven't tried it, I'm pretty 
sure the Linux USB audio driver will probably see it.


-Steve

John covici wrote:

Any way to use this on a Linux box so I could use this with asterisk?
I have a windows box on the same network, but how would I get asterisk
to see such a thing?

Thanks.

on Wednesday 04/26/2006 Steve Feinstein([EMAIL PROTECTED]) wrote
  http://www.iogear.com/main.php?loc=productItem=GPH100U
  
  I've got a couple of these, they're $40 and there's a $20 rebate going 
  on now. For that price it's pretty amazing. Plug and play, no drivers 
  required. Quality very good, it does echo cancellation and noise 
  reduction. I only wish it had a mute button.
  
  BTW: I don't have any affiliation with ioGear other than I like this 
  product.
  
  
  Jim Houser wrote:

   Personal preference. I'm not a big headset guy.
   The real point of my reply was to say how impressed I am with USB talk 
   quality when compared to a hardphone on Asterisk or our Avaya 
   Communications Manager. Like my wife says, I guess I'm not being 
   clear... :)

  
   
   *From:* [EMAIL PROTECTED] 
   [mailto:[EMAIL PROTECTED] *On Behalf Of *Dean 
   Collins

   *Sent:* Wednesday, April 26, 2006 10:24 AM
   *To:* Asterisk Users Mailing List - Non-Commercial Discussion
   *Subject:* RE: [Asterisk-Users] USB conference phone
  
   Kerry, do you actually own one? Have you used it for long? What are 
   you using it for?

  
   (jim ? personally I cant see the point of using your phone when I have 
   a very good quality headset and mic.).

  
   Dean
  
   
  
   *From:* [EMAIL PROTECTED] 
   [mailto:[EMAIL PROTECTED] *On Behalf Of *Kerry 
   Garrison

   *Sent:* Wednesday, 26 April 2006 10:36 AM
   *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
   *Subject:* RE: [Asterisk-Users] USB conference phone
  
   This is an excellent USB speakerphone
  
   http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 
   http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27

  
   

  
   *From:* [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] *On Behalf Of
   *Jim Houser
   *Sent:* Wednesday, April 26, 2006 6:26 AM
   *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
   *Subject:* RE: [Asterisk-Users] USB conference phone
  
   I don't know about this phone but I can tell you I have a vendor
   that will only talk to me via Skype so I purchased this:
   http://www.provantage.com/usb-internet-phone~220150620.htm
   http://www.provantage.com/usb-internet-phone%7E220150620.htm
  
   It operates nice and has very good call quality.
  
   

  
   *From:* [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] *On Behalf Of
   *Dean Collins
   *Sent:* Tuesday, April 25, 2006 8:22 PM
   *To:* Asterisk Users Mailing List - Non-Commercial Discussion
   *Subject:* [Asterisk-Users] USB conference phone
  
   Has anyone actually used these USB speakerphones
  
   
http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
  
   Seems to get a pretty good review here
  
   
http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27
   
http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27
  
   But looking for real world feedback.
  
   Cheers,
  
   Dean
  
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   immediately by return e-mail and destroy any copies. Any
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Re: [Asterisk-Users] USB conference phone

[Steve Feinstein]

http://www.iogear.com/main.php?loc=productItem=GPH100U

I've got a couple of these, they're $40 and there's a $20 rebate going 
on now. For that price it's pretty amazing. Plug and play, no drivers 
required. Quality very good, it does echo cancellation and noise 
reduction. I only wish it had a mute button.


BTW: I don't have any affiliation with ioGear other than I like this 
product.



Jim Houser wrote:

Personal preference. I'm not a big headset guy.
The real point of my reply was to say how impressed I am with USB talk 
quality when compared to a hardphone on Asterisk or our Avaya 
Communications Manager. Like my wife says, I guess I'm not being 
clear... :)



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Dean 
Collins

*Sent:* Wednesday, April 26, 2006 10:24 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [Asterisk-Users] USB conference phone

Kerry, do you actually own one? Have you used it for long? What are 
you using it for?


(jim – personally I cant see the point of using your phone when I have 
a very good quality headset and mic.).


Dean



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Kerry 
Garrison

*Sent:* Wednesday, 26 April 2006 10:36 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* RE: [Asterisk-Users] USB conference phone

This is an excellent USB speakerphone

http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 
http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27




*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Jim Houser
*Sent:* Wednesday, April 26, 2006 6:26 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* RE: [Asterisk-Users] USB conference phone

I don't know about this phone but I can tell you I have a vendor
that will only talk to me via Skype so I purchased this:
http://www.provantage.com/usb-internet-phone~220150620.htm
http://www.provantage.com/usb-internet-phone%7E220150620.htm

It operates nice and has very good call quality.



*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Dean Collins
*Sent:* Tuesday, April 25, 2006 8:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [Asterisk-Users] USB conference phone

Has anyone actually used these USB speakerphones


http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem

Seems to get a pretty good review here

http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27

http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27

But looking for real world feedback.

Cheers,

Dean

This e-mail and any attachments may contain confidential and
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please notify the sender, or [EMAIL PROTECTED],
immediately by return e-mail and destroy any copies. Any
dissemination or use of this information by a person other than
the intended recipient is unauthorized and may be illegal. Unless
otherwise stated, opinions expressed in this e-mail are those of
the author and are not endorsed by the author's employer.

This e-mail and any attachments may contain confidential and 
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email;internet:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP

[Steve Feinstein]

Actually it makes no difference.  I tried it in an attempt to get 
something to happen.


Thanks,
-Steve

Eric ManxPower Wieling wrote:

What happens if you remove the r option?  r is almost NEVER useful.

Steve Feinstein wrote:

I've been pulling my hair out over this one trying to understand it.

If you have a very simple extension:

exten = 1,n,Dial(IAX2/Steve|24|r)

Everything I've seen says this should tell the IAX phone (our own 
iaxclient based one) to make a ringing sound, or asterisk should make 
the ringback indication itself if it determines that the channel 
can't do it for itself.


But you can dial this extension all day and you never hear a ringback 
indication.  Dial it from a SIP softphone and you do.  If you change 
the default country in the indications.conf, the SIP phone will 
change the way the ring sounds.  IAX, still nothing.


You can use PlayTones(ring) in the dialplan before the Dial(), and it 
seems to behave ok.  Playing the appropriate ring indication until 
the call is answered.  But it seems like the behavior is inconsistent 
with IAX vs. SIP.  Is this by design?


All the IAX soft phones I've tried are based on the same iaxclient 
libs, so it's hard to know if it's the phone or asterisk that's not 
behaving right.  Has anyone used an iax hard phone, some other IAX 
device/software, and does it exhibit the same behavior?  Or is this a 
problem with the iax code not being telling asterisk that IAX phones 
need to have their indications faked.


Any ideas about what's going on would be most gratefully appreciated.



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Re: [Asterisk-Users] MixMonitor and filenames

[Steve Feinstein]

Seems like mixmonitor app uses the extension to determine what format to 
save the file as.  (ie raw, gsm, etc.)So I think you want to leave 
the extension alone.  But you have lot's of control over the filename 
itself using variables  functions.  Here's what I'm doing which seems 
marginally useful.  It saves every file with the channel name, and the 
unique ID as the filename.  The URI encode is because the channel name 
probably has a slash in it.


exten = StartTest,n,MixMonitor(${URIENCODE(${CHANNEL})}${UNIQUEID}.gsm)

Now all I need is a way to play them back one at a time  delete them  
like voicemails.  I guess I could save them to a voicemail directory 
structure and name them so that voicemail app would be happy with them.  
But I'd run in to the same problem that there doesn't seem to be a way 
to read a directory from a dialplan app or function.  I supposed there's 
always AGI for that.  But this was supposed to be quick and dirty.


Eric Jacksch wrote:

A client wants to record all calls to a specific extension.  MixMonitor
seems to do the job, but is there a way to get it to append something to the
filename for each call?  Right now it overwrites the file every time a call
comes in.

I realize there is an append option, but I'd prefer a separate file per
call.

Thanks,
Eric

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[Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP

[Steve Feinstein]

I've been pulling my hair out over this one trying to understand it.

If you have a very simple extension:

exten = 1,n,Dial(IAX2/Steve|24|r)

Everything I've seen says this should tell the IAX phone (our own 
iaxclient based one) to make a ringing sound, or asterisk should make 
the ringback indication itself if it determines that the channel can't 
do it for itself.


But you can dial this extension all day and you never hear a ringback 
indication.  Dial it from a SIP softphone and you do.  If you change the 
default country in the indications.conf, the SIP phone will change the 
way the ring sounds.  IAX, still nothing.


You can use PlayTones(ring) in the dialplan before the Dial(), and it 
seems to behave ok.  Playing the appropriate ring indication until the 
call is answered.  But it seems like the behavior is inconsistent with 
IAX vs. SIP.  Is this by design?


All the IAX soft phones I've tried are based on the same iaxclient libs, 
so it's hard to know if it's the phone or asterisk that's not behaving 
right.  Has anyone used an iax hard phone, some other IAX 
device/software, and does it exhibit the same behavior?  Or is this a 
problem with the iax code not being telling asterisk that IAX phones 
need to have their indications faked.


Any ideas about what's going on would be most gratefully appreciated.

-Steve Feinstein  (asterisk 1.2.7.1 btw)
GatherWorks, Inc.

begin:vcard
fn:Steve Feinstein
n:Feinstein;Steve
org:GatherWorks Inc.
adr:;;27 Dupaw Gould Road;Brookline;NH;03033;USA
email;internet:[EMAIL PROTECTED]
tel;work:+1 (603) 672-1472
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[Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone

[Steve Feinstein]

I'd like for our custom soft phone to be able to know what queue, and/or 
what DID is calling an Agent's phone before the agent picks up.  The 
agent is using the AGENTCALLBACKLOGIN.  One agent can be in multiple 
queues so it would be nice if they could get a pop up window telling 
them who's on the line before they pick up and hear the announcement 
telling them that.  I'd like to lose the announcment all together.


It seems like that the phone can easily know what extension was dialed 
to make it ring, but at best that's the phone client's extension (The 
server dialed it via the Local/ interface), and at worst it's 's'.  Is 
there anyway I can know the DID of the person who called into the Queue?


I've done ethereal traces and it seems like the DID, that actually 
called the agent/phone is no where to be found. 
I've tried also to use the URL string in the Queue() application, but 
the server doesn't seem to send it.  (I've also tried having the client 
send a URL, and it seems to get sent, yet the server doesn't seem to 
forward it.  It seems to just get lost). 

Has anyone gotten the URL in the Queue application to work?  And if it 
does, it it delivered to the phone before, or after the phone answers?


Any hacks,tips,tricks,pointers, would be most appreciated.

Thanks,
Steve Feinstein
GatherWorks Inc.

begin:vcard
fn:Steve Feinstein
n:Feinstein;Steve
org:GatherWorks Inc.
adr:;;27 Dupaw Gould Road;Brookline;NH;03033;USA
email;internet:[EMAIL PROTECTED]
tel;work:+1 (603) 672-1472
x-mozilla-html:TRUE
url:http://www.gatherworks.com/
version:2.1
end:vcard

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Re: [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone

[Steve Feinstein]

Thanks!, I will definitely take a look at that.  We were hoping not to 
have to do AGI in the client, but if we have to, we have to.  It'll 
probably be useful for other things down the road.


-Steve Feinstein
GatherWorks Inc.

BJ Weschke wrote:

On 4/12/06, Steve Feinstein [EMAIL PROTECTED] wrote:
  

I'd like for our custom soft phone to be able to know what queue, and/or
what DID is calling an Agent's phone before the agent picks up.  The
agent is using the AGENTCALLBACKLOGIN.  One agent can be in multiple
queues so it would be nice if they could get a pop up window telling
them who's on the line before they pick up and hear the announcement
telling them that.  I'd like to lose the announcment all together.

It seems like that the phone can easily know what extension was dialed
to make it ring, but at best that's the phone client's extension (The
server dialed it via the Local/ interface), and at worst it's 's'.  Is
there anyway I can know the DID of the person who called into the Queue?

I've done ethereal traces and it seems like the DID, that actually
called the agent/phone is no where to be found.
I've tried also to use the URL string in the Queue() application, but
the server doesn't seem to send it.  (I've also tried having the client
send a URL, and it seems to get sent, yet the server doesn't seem to
forward it.  It seems to just get lost).

Has anyone gotten the URL in the Queue application to work?  And if it
does, it it delivered to the phone before, or after the phone answers?

Any hacks,tips,tricks,pointers, would be most appreciated.




http://bugs.digium.com/view.php?id=6843

 Here's code to fire off an AGI to do pretty much anything you need to
do on the calling channel after a Queue Member has been assigned to
it.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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begin:vcard
fn:Steve Feinstein
n:Feinstein;Steve
org:GatherWorks Inc.
adr:;;27 Dupaw Gould Road;Brookline;NH;03033;USA
email;internet:[EMAIL PROTECTED]
tel;work:+1 (603) 672-1472
x-mozilla-html:TRUE
url:http://www.gatherworks.com/
version:2.1
end:vcard

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Re: [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone

[Steve Feinstein]

Kyle,
That's bloody brilliant

Thanks so much!

-Steve Feinstein
GatherWorks, Inc.

Kyle Sexton wrote:

Have you tried something like:

exten = 2,1,SetCIDName(QUEUENAME: ${CALLERIDNAME})
exten = 2,n,Queue(QUEUENAME)





On 4/12/06, * Steve Feinstein* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Thanks!, I will definitely take a look at that.  We were hoping not to
have to do AGI in the client, but if we have to, we have to.  It'll
probably be useful for other things down the road.

-Steve Feinstein
GatherWorks Inc.

BJ Weschke wrote:
 On 4/12/06, Steve Feinstein [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:

 I'd like for our custom soft phone to be able to know what
queue, and/or
 what DID is calling an Agent's phone before the agent picks
up.  The
 agent is using the AGENTCALLBACKLOGIN.  One agent can be in
multiple
 queues so it would be nice if they could get a pop up window
telling
 them who's on the line before they pick up and hear the
announcement
 telling them that.  I'd like to lose the announcment all together.

 It seems like that the phone can easily know what extension was
dialed
 to make it ring, but at best that's the phone client's
extension (The
 server dialed it via the Local/ interface), and at worst it's
's'.  Is
 there anyway I can know the DID of the person who called into
the Queue?

 I've done ethereal traces and it seems like the DID, that actually
 called the agent/phone is no where to be found.
 I've tried also to use the URL string in the Queue()
application, but
 the server doesn't seem to send it.  (I've also tried having
the client
 send a URL, and it seems to get sent, yet the server doesn't
seem to
 forward it.  It seems to just get lost).

 Has anyone gotten the URL in the Queue application to
work?  And if it
 does, it it delivered to the phone before, or after the phone
answers?

 Any hacks,tips,tricks,pointers, would be most appreciated.



 http://bugs.digium.com/view.php?id=6843

  Here's code to fire off an AGI to do pretty much anything you
need to
 do on the calling channel after a Queue Member has been assigned to
 it.

 --
 Bird's The Word Technologies, Inc.
 http://www.btwtech.com/
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begin:vcard
fn:Steve Feinstein
n:Feinstein;Steve
org:GatherWorks Inc.
adr:;;27 Dupaw Gould Road;Brookline;NH;03033;USA
email;internet:[EMAIL PROTECTED]
tel;work:+1 (603) 672-1472
x-mozilla-html:TRUE
url:http://www.gatherworks.com/
version:2.1
end:vcard

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