from:"Kevin Savoy"

RE: [asterisk-users] balance anouncement

2006-09-01 Thread Kevin Savoy









Its telling you the sound file Goodbye
does not exist in the directory it looks for sounds. If you indeed have a sound
file called Goodbye then you need to either move it to the default sounds
directory or add the path line to the command. If you dont have the
sound file youll need to either create one or use one you do have.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ram
Sent: Friday, September 01, 2006
11:06 AM
To:
[EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
Subject: Re: [asterisk-users]
balance anouncement







Hi











iam trying like this in my extension.conf





some one refered in the news group past

















error in messages











Sep 1 21:31:42 WARNING[28610] file.c: File Goodbye does not exist
in any format
Sep 1 21:31:42 WARNING[28610] file.c: Unable to open Goodbye (format
ulaw): No such file or directory
Sep 1 21:31:42 WARNING[28610] app_playback.c: ast_streamfile failed on
SIP/8-3ca6 for Goodbye 


exten = 888,1,Read(${CALLERIDNUM})
exten = 888,2,MYSQL(Connect connid 127.0.0.1
root password mydatabase)
exten = 888,3,MYSQL(Query resultid ${connid} selecttotal from balance
where username=${CALLERIDNUM}) 
exten = 888,4,MYSQL(Fetch fetch ${resultid} AMOUNT-DUE)
exten = 888,5,MYSQL(Clear ${resultid})
exten = 888,6,Playback(Goodbye)
exten = 888,7,Hangup

















iam calling from my extension to 888











Ram












On 9/1/06, John
Millican [EMAIL PROTECTED]
wrote: 

On Friday September 01 2006 10:19 am, ram wrote:
 Hi

 thanks for the quick reply

 any documents to read to achive this
 or any examples would be great to read

 Ram

 On 9/1/06, John Millican [EMAIL PROTECTED]
 wrote:
  On Friday September 01 2006 9:27 am, ram wrote:
   Hi
  
   how can i do balance anouncement by using asterisk
  
   take example, i have table balance , user name 9, balance
200$ 
  
   user dial *98 or what ever, then i need anouce his balance is
200$, by
   reading from that row
  
   any clues how can i achive this or is this possible ? 
  
   Ram
 
  Create an AGI script that does a db look up for the ballance and then
  pass the
  result back to Cepstral or Festival or your favorite text to speech 
  software.
  John M
 
Try google or voip-info.org and search for
Asterisk AGIshould yeid some good results.
AGI can be called from the dial plan and written in your favorite language 
i.e. PHP, C++, Perl, C, Java
or start here:
http://home.cogeco.ca/~camstuff/agi.html
http://asterisk.drunkcoder.com/agi.cgi


John M

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RE: [asterisk-users] PRI and Asterisk

2006-08-23 Thread Kevin Savoy









I have tested Redfones boxes. Tried
two of them and was able to re-create some issues. I did not have PRI lines but
a 24 channel em wink line so not sure if PRI is affected as well. I found
that over time we had issues with hanging zap channels. Asterisk reported
everything was just fine yet people got busy signals calling in and when
calling out all they got was silence. The CLI never showed any incoming calls
that were attempted and when dialing out it showed Dialing but nothing
happened. I worked with Mark Warren at Redfone and he was very co-operative and
had an idea to fix this but sadly we just didnt have any more time to
fight with it and went with Digium cards. As of this writing I am starting to
get problems with inbound calls. Seems for a couple minutes no one can dial
into our office and then it just clears up. No errors or anything in Asterisk
to indicate a problem. 











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian Varanini
Sent: Tuesday, August 22, 2006
6:35 PM
To: Julian Varanini
Subject: RE: [asterisk-users] PRI
and Asterisk





Hi Everyone

Any opinions on this?

Thanks

Julian












From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Tue, 18 Jul 2006 01:29:57 +
Subject: [asterisk-users] PRI and Asterisk

Hi All,


I am planning to order a PRI and would like to know your opinions on a devices
like the Redfone redbridge. Basically any PRI to Asterisk interface that has
worked well for you.

Thanks,

Julian






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[asterisk-users] Unable to start special tone

2006-08-23 Thread Kevin Savoy









Can
anyone tell me where this is coming from? I cant seem to find any information
on it anywhere. I dont believe Im using special tones
anywhere. Any ideas?



Aug 23
14:30:34 WARNING[15443]: chan_zap.c:5492 ss_thread: Unable to start special
tone on 15



_

Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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[asterisk-users] Text to Speech

2006-08-21 Thread Kevin Savoy









Can
someone recommend a good text to speech engine that is usable by Asterisk? I
have tried the Festival one and it just doesnt cut it for commercial
applications. 



We
are willing to pay for a good one that works. Anyone tried the ATT speech
engine? The IBM ViaVoice sounds no better then Festival.



Thanks
for your input.



_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [asterisk-users] Text to Speech

2006-08-21 Thread Kevin Savoy

All I can find for Flite is for AAH, does it work as well with plain
Asterisk? Is the setup the same?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Monday, August 21, 2006 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Text to Speech

N.B.: Please use plain text when sending to this list

 Can someone recommend a good text to speech engine that is usable by
Asterisk? I have tried the Festival one and it just doesn't cut it for
commercial applications.

 We are willing to pay for a good one that works. Anyone tried the ATT
speech engine? The IBM ViaVoice sounds no better then Festival.

You have flite that is free and, IMHO better than festival
(http://nerdvittles.com/index.php?p=134).

I also tried Prophecy (http://www.voxeo.com/) but I'm still waiting
for the Linux version as I don't have time to babysit a Windows server
:)

hth
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RE: [asterisk-users] Auto retry on Busy

2006-08-11 Thread Kevin Savoy

You need to change:

exten = 777,4,goto(trunkretry,1,1)

to

exten = 777,4,goto(trunkretry,777,1)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Silverman
Sent: Friday, August 11, 2006 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Kevin Smith
Subject: Re: [asterisk-users] Auto retry on Busy


Kevin,

Thanks for the suggestion.  I can't seem to get it to work.

This is what I put in my extensions.conf

We only have one number that we want to keep trying right now, so I  
tried to set it so by calling extension 777, it would start the  
system retrying.  (The actual number isn't 999 :)

[trunkretry]
exten = 777,1,Dial(${TRUNK}/www1323999},10,)
exten = 777,2,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
exten = 777,3,(LINEBUSY), Wait(15)
exten = 777,4,goto(trunkretry,1,1)


Thanks,

-N



On Aug 11, 2006, at 11:29 AM, Kevin Smith wrote:

 Why don't you just test for the dial status after the dial command  
 completes? I don't really see why you want something to keep  
 dialing until it gets through, but this would work.

 [something]
 1,1,Dial(zap/,sip/, etc/whatever, 10)
 1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
 1,n(LINEBUSY), Wait(30)
 1,n,goto(something,1,1)
 1,n(OTHER), do something else

 Sure it is pretty rough, but the basics are there. Also you might  
 want to read this: http://www.voip-info.org/wiki-Asterisk+variable 
 +DIALSTATUS

 Kevin



 Noah Silverman wrote:
 Hi,

 Does anybody have an easy solution for this.

 I want something that will keep trying a busy number every 30  
 seconds until it gets through.

 I've tried retrydial, but can't get it to work.

 Any suggestions?

 Thanks,

 -N
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RE: [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk

2006-08-10 Thread Kevin Savoy

This is an issue I'm having as well. Here is what I've discovered. 

Call comes in on a T1 line. Call is sent to a SIP phone (say 4000) based on
the extensions.conf setup. User of phone 4000 has set a forward in the phone
to an external number, 1-555-555-. There is nothing telling Asterisk to
Dial(Zap/g1) so the call does not get converted back to slin to send along
the T1 lines out of the building. Since SIP can't be sent the frame is
incompatible and is dropped. I know this probably isn't as technical as it
should be but in essence it is what is happening. I've had to do a
workaround and set up an extension that dials the number that the phone was
to be forwarded too. I set up extension 500. The user forwards the phone to
500. extensions.conf says Dial(Zap/g1/155).

Band-aid solution. I've seen on the bug reports it is a known issue but not
resolved yet. Last update was July 5th.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of M D
Sent: Thursday, August 10, 2006 8:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk

Hi there

We're running Asterisk 1.2.1 (I know, it's old; we have an upgrade
planned but can't do it just yet) on Debian testing. Every now and
Asterisk and the box are dying -- no SSH login, no calls, nothing. The
last lines logged are:

Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Executing
Dial(SIP/5060-0843a7f0, SIP/123456|30)
Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Called 123456
Jul 31 14:23:31 VERBOSE[14085] logger.c: -- Got SIP response 302
Moved Temporarily back from 85.189.x.x
Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Now forwarding
SIP/5060-0843a7f0 to 'Local/[EMAIL PROTECTED]' (thanks to
SIP/123456-2241)
Jul 31 14:23:31 VERBOSE[32701] logger.c: -- Executing
Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]:5070) in new stack
Jul 31 14:23:31 VERBOSE[32701] logger.c : -- Called
[EMAIL PROTECTED]:5070
Jul 31 14:23:31 VERBOSE[32701] logger.c: --
SIP/outbound.gateway:5070-550a is ringing
Jul 31 14:23:31 VERBOSE[32696] logger.c: --
Local/[EMAIL PROTECTED],1 is ringing
Jul 31 14:23:31 VERBOSE[32701] logger.c: --
SIP/outbound.gateway:5070-550a is making progress passing it to
Local/[EMAIL PROTECTED],2
Jul 31 14:23:31 VERBOSE[32696] logger.c: --
Local/[EMAIL PROTECTED] _110-7282,1 is making progress passing it to
SIP/5060-0843a7f0
Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
frame on Local/[EMAIL PROTECTED],2 of format slin since our
native format has changed to alaw
Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
frame on Local/[EMAIL PROTECTED],2 of format slin since our
native format has changed to alaw

The last lines are repeated until the server dies.

The phone appears to be a SNOM and should be using only g.711 alaw or ulaw.

I inherited this box with Asterisk running as root so I've changed it
to a non-privileged user but assuming the server is dynig through
resource starvation I doubt it'll help.

So, any ideas what this traffic is? What can we do to stop it? Clearly
I need to upgrade Asterisk but a cursory glance at the changelog
doesn't suggest a bug was reported with these symptoms which would
have been fixed in a later release.

Cheers,

Mark
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RE: [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk

2006-08-10 Thread Kevin Savoy

My Asterisk is 1.2.9.1 but I've recreated this on 1.2.7 and 1.2.8. Not tried
1.2.10 yet.

This only happens on forwarded calls for me as well. I've not let it run too
long to see if the server dies eventually. I don't believe it will because
once the caller hangs up the errors stop and my server carries on. I've not
had it die on me yet but I've no told our users NEVER to forward outside the
building.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of M D
Sent: Thursday, August 10, 2006 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fwd: Dropping incompatible frame killing
Asterisk

Hi

Sorry, I should have mentioned that we're only running SIP. Our calls
to the PSTN are routed through a VoIP carrier and all of our clients
are SIP.

Which version of Asterisk are you using? Is this killing your box? If
it is, have you established why? CPU being killed, memory starvation,
something else?

It is only happening on forwarded calls, though. I'll have to try your
workaround.

Thanks,

Mark

On 10/08/06, Kevin Savoy [EMAIL PROTECTED] wrote:
 This is an issue I'm having as well. Here is what I've discovered.

 Call comes in on a T1 line. Call is sent to a SIP phone (say 4000) based
on
 the extensions.conf setup. User of phone 4000 has set a forward in the
phone
 to an external number, 1-555-555-. There is nothing telling Asterisk
to
 Dial(Zap/g1) so the call does not get converted back to slin to send along
 the T1 lines out of the building. Since SIP can't be sent the frame is
 incompatible and is dropped. I know this probably isn't as technical as it
 should be but in essence it is what is happening. I've had to do a
 workaround and set up an extension that dials the number that the phone
was
 to be forwarded too. I set up extension 500. The user forwards the phone
to
 500. extensions.conf says Dial(Zap/g1/155).

 Band-aid solution. I've seen on the bug reports it is a known issue but
not
 resolved yet. Last update was July 5th.


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of M D
 Sent: Thursday, August 10, 2006 8:50 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Fwd: Dropping incompatible frame killing
Asterisk

 Hi there

 We're running Asterisk 1.2.1 (I know, it's old; we have an upgrade
 planned but can't do it just yet) on Debian testing. Every now and
 Asterisk and the box are dying -- no SSH login, no calls, nothing. The
 last lines logged are:

 Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Executing
 Dial(SIP/5060-0843a7f0, SIP/123456|30)
 Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Called 123456
 Jul 31 14:23:31 VERBOSE[14085] logger.c: -- Got SIP response 302
 Moved Temporarily back from 85.189.x.x
 Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Now forwarding
 SIP/5060-0843a7f0 to 'Local/[EMAIL PROTECTED]' (thanks to
 SIP/123456-2241)
 Jul 31 14:23:31 VERBOSE[32701] logger.c: -- Executing
 Dial(Local/[EMAIL PROTECTED],2,
 SIP/[EMAIL PROTECTED]:5070) in new stack
 Jul 31 14:23:31 VERBOSE[32701] logger.c : -- Called
 [EMAIL PROTECTED]:5070
 Jul 31 14:23:31 VERBOSE[32701] logger.c: --
 SIP/outbound.gateway:5070-550a is ringing
 Jul 31 14:23:31 VERBOSE[32696] logger.c: --
 Local/[EMAIL PROTECTED],1 is ringing
 Jul 31 14:23:31 VERBOSE[32701] logger.c: --
 SIP/outbound.gateway:5070-550a is making progress passing it to
 Local/[EMAIL PROTECTED],2
 Jul 31 14:23:31 VERBOSE[32696] logger.c: --
 Local/[EMAIL PROTECTED] _110-7282,1 is making progress passing it to
 SIP/5060-0843a7f0
 Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
 frame on Local/[EMAIL PROTECTED],2 of format slin since our
 native format has changed to alaw
 Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
 frame on Local/[EMAIL PROTECTED],2 of format slin since our
 native format has changed to alaw

 The last lines are repeated until the server dies.

 The phone appears to be a SNOM and should be using only g.711 alaw or
ulaw.

 I inherited this box with Asterisk running as root so I've changed it
 to a non-privileged user but assuming the server is dynig through
 resource starvation I doubt it'll help.

 So, any ideas what this traffic is? What can we do to stop it? Clearly
 I need to upgrade Asterisk but a cursory glance at the changelog
 doesn't suggest a bug was reported with these symptoms which would
 have been fixed in a later release.

 Cheers,

 Mark
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[asterisk-users] Operator in Voicemail

2006-07-24 Thread Kevin Savoy









Ive
got an odd problem. I have set in Voicemail.conf operator=yes as a default. This
is so that when a caller is in the voicemail system they can press 0 and be
sent to the operator. This works fine when the caller is internal to the system
but NOT when the caller is calling in from the PSTN. Instead the caller gets
the message Press 1 to accept the recording. Pressing 0 again deletes the
message. How do I get this to work for outside callers calling in??



Thanks







_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [asterisk-users] Operator in Voicemail

2006-07-24 Thread Kevin Savoy









This doesnt solve the problem. Still the same. Any
other ideas?











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henk
Sent: Monday, July 24, 2006 12:25
PM
To: 'Asterisk Users Mailing List -
 Non-Commercial Discussion'
Subject: RE: [asterisk-users]
Operator in Voicemail





This is what I am using:



exten = o,1,Answer()

exten = o,2,GoTo(default,3000,1)

exten = o,3,Hangup()



Hope this helps,



Henk











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Davis
Sent: maandag 24 juli 2006 18:20
To: Asterisk Users Mailing List -
 Non-Commercial Discussion
Subject: RE: [asterisk-users]
Operator in Voicemail





Im having the exact same problem here. I originally
thought it was a context problem. 

However, to troubleshoot I tried placing the following in
every context (default, from-inside, from-outside, etc) in extensions.conf with
no luck:

exten =
o,1,DIAL(SIP/100,100)



Like Kevin, it works fine for our internal users, just
doesnt work for callers coming from the PSTN.



Thanks,

-AntD









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy
Sent: Monday, July 24, 2006 7:37
AM
To: 'Asterisk Users Mailing List -
 Non-Commercial Discussion'
Subject: [asterisk-users] Operator
in Voicemail





Ive
got an odd problem. I have set in Voicemail.conf operator=yes as a default.
This is so that when a caller is in the voicemail system they can press 0 and
be sent to the operator. This works fine when the caller is internal to the
system but NOT when the caller is calling in from the PSTN. Instead the caller
gets the message Press 1 to accept the recording. Pressing 0 again deletes the
message. How do I get this to work for outside callers calling in??



Thanks







_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [asterisk-users] Operator in Voicemail

2006-07-24 Thread Kevin Savoy

It is an o as in operator. That's what the manuals say. O extension is
operator.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, July 24, 2006 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Operator in Voicemail

Are you sure this is saying exten = 0 with a ZERO and not an Oh?
Looks like a lowercase Oh to me below.

Kevin Savoy wrote:
 This doesn't solve the problem. Still the same. Any other ideas?

 This is what I am using:

 exten = o,1,Answer()

 exten = o,2,GoTo(default,3000,1)

 exten = o,3,Hangup()

 Hope this helps,

 Henk

 *From:* [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] *On Behalf Of *Anthony 
 Davis
 *Sent:* maandag 24 juli 2006 18:20
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* RE: [asterisk-users] Operator in Voicemail

 I'm having the exact same problem here. I originally thought it was a 
 context problem.

 However, to troubleshoot I tried placing the following in every context 
 (default, from-inside, from-outside, etc) in extensions.conf with no luck:

  / exten = o,1,DIAL(SIP/100,100)/

 Like Kevin, it works fine for our internal users, just doesn't work for 
 callers coming from the PSTN.

 Thanks,

 -AntD

 *From:* [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] *On Behalf Of *Kevin
Savoy
 *Sent:* Monday, July 24, 2006 7:37 AM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* [asterisk-users] Operator in Voicemail

 I've got an odd problem. I have set in Voicemail.conf operator=yes as a 
 default. This is so that when a caller is in the voicemail system they 
 can press 0 and be sent to the operator. This works fine when the caller 
 is internal to the system but NOT when the caller is calling in from the 
 PSTN. Instead the caller gets the message Press 1 to accept the 
 recording. Pressing 0 again deletes the message. How do I get this to 
 work for outside callers calling in??

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RE: [asterisk-users] Email notification of voicemail

2006-07-17 Thread Kevin Savoy

Sheesh, tried that and I'm STILL getting email attempts. This doesn't make
sense here. Any other ideas?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Friday, July 14, 2006 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Email notification of voicemail

Aha - get rid of the leading comma for each entry..

 = ,Front Desk
 = ..

A.

On Jul 13, 2006, at 1:00 PM, Kevin Savoy wrote:

 I've X'd out the extensions and passwords but this is all I have in 
 there.
 Thanks

 [default]
 =,,Front Desk,,

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RE: [asterisk-users] Voicemail CallerID

2006-07-13 Thread Kevin Savoy

Would this work?
exten = 3299,1,VoicemailMain(${EXTEN})

This way it would check the voicemail of the extension doing the dialing?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peder @
NetworkOblivion
Sent: Thursday, July 13, 2006 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail  CallerID

I've got a question about voicemail and callerid and I can't quite 
figure it out. I've got extensions 100, 101 and 102.  For outbound 
callerID (calls from the phones to the PSTN), I want the callerid to say 
100 on all phones, so under sip.conf, I added:

callerid=Bill 100

The problem is that when they go to check voicemail, it looks at their 
callerID and it drops them into mailbox 100 (calls to them still go into 
their own specific mailbox, it is just when they hit their messages 
button).  Any idea how to get around that?  Or do I just have to send 
them to voicemail without having it automatically enter their extension?

This is what my voicemail does:
exten = 3299,1,VoicemailMain(${CALLERIDNUM})

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RE: [asterisk-users] quad T1 pri

2006-07-13 Thread Kevin Savoy

Should be span 1 for the for T1 and span 2 for the second T1 in your config.
They are both span 1.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Thursday, July 13, 2006 12:47 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] quad T1 pri

I am connecting my Quad T1 card to the Siemens HIpath 4000 with pri.
Port 1 is working just fine. But port 2 is not working.
I think I have my configuration correct (see below).
Is there something special about configuring 2 PRI? I have done it with 
dual T1 no problem.
I am using asterisk 1.2.9.1 libpri-1.2.3 and zaptel 1.2.6.

THanks,

Jerry

--
zaptel.conf:

span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=1,1,0,esf,b8zs
bchan=25-47
dchan=48

zapata.conf:
signalling=pri_cpe
switchtype=national
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
context=smvoice-incoming
group=1
channel = 1-23

signalling=pri_cpe
switchtype=national
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
context=smvoice-incoming
group=1
channel = 25-47

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RE: [asterisk-users] Email notification of voicemail

2006-07-13 Thread Kevin Savoy

Thanks for replying. Have tried that. If I don't specify an email address it
then takes the first name and last name and then the domain of the pbx. For
example 

1234 = 1234,Bob Smith

I then get:

[EMAIL PROTECTED]

Which of course fails because that address doesn't exist.

Any other ideas?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Thursday, July 13, 2006 2:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Email notification of voicemail

Try having nothing after the name in your voicemail.conf:

1234 = 1234,The Marquis de Sade

Regards,
--  
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Jul 12, 2006, at 11:17 AM, Kevin Savoy wrote:

 I have attach=no in my voicemail.conf so that can't be doing it. Not  
 sure
 where that sendmail command is. Don't see it in voicemail.conf or any  
 other
 config in the asterisk directory.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of VoIP  
 Street
 Sent: Wednesday, July 12, 2006 12:01 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Email notification of voicemail

 Kevin Savoy wrote:
  Asterisk is trying to send an email to users when they receive a
  voicemail. Can this be shut off? I have not entered any email  
 addresses
  in voicemail.conf so it tries to send to  
 [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED]. This of course gets  
 rejected
  since the user does not exist and the root users mailbox on linux  
 gets
  full of these rejection notices. I can't seem to find anywhere to  
 tell
  Asterisk to stop notifying people they have voicemails.
 
  
 
  I'm using 1.2.9.1 of Asterisk. Thanks
 
  
 
  _
 
  
 
  **Kevin Savoy**
 
  **Business Unit Telecom Analyst**
 
  2218 4th Ave W
 
  Williston, ND 58801
 
  Ph: 701-774-4023
 
  Fax: 701-774-2901
 
  http://www.novo1.com
 
  Novo 1 is a service mark of Novo 1, Inc
 
  
 
 
   
 --- 
 -
 
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 You could try commenting out:

 attach=yes

 Also, if you don't want any emails sent ever for any voice mail users
 you could probably uncomment the following line and give it a bogus  
 path
 to the mailer.

 ;mailcmd=/usr/sbin/sendmail -t

 There is probably a better way to do this but we have never needed to
 turn it off so I am not sure.

 Hope this helps.

 -- 
 VoIP Street
 Origination/Termination with SUPERIOR customer service!
 http://www.VoIPstreet.com
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RE: [asterisk-users] Email notification of voicemail

2006-07-13 Thread Kevin Savoy

I've X'd out the extensions and passwords but this is all I have in there.
Thanks 

[general]
attach=no
format=wav49
skipms=3000
nextaftercmd=yes
maxsilence=10
silencethreshold=128
maxlogins=3
tz=central
operator=yes

[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/CST6CDT|'vm-received' Q 'digits/at' IMp
mountain=America/Phoenix|'vm-received' q 'digits/at' IMp
pacific=America/Los_Angeles|'vm-received' q 'digits/at' IMp
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'

[default]
=,,Front Desk,,
=,,Eric Kelley
=,,Justin Hall
=,,Jason Smestad
=,,Cumi Everson
=,,Glenda Cusker
=,,Laura Sanford
=,,Gary Sundet
=,,Kevin Penner
=,,Kevin Savoy
=,,Jeff Garaas
=,,Natalie Thompson
=,,Jolene Ross
=,,Ralf Patterson
=,,Mike Satterlee
=,,Michelle Siverson
=,,Kathy Evenson
=,,Kathy Michels

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Thursday, July 13, 2006 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Email notification of voicemail

Can you send me (or pastebin) your voicemail.conf?

A.

On Jul 13, 2006, at 12:35 PM, Kevin Savoy wrote:

 Thanks for replying. Have tried that. If I don't specify an email 
 address it
 then takes the first name and last name and then the domain of the 
 pbx. For
 example

 1234 = 1234,Bob Smith

 I then get:

 [EMAIL PROTECTED]

 Which of course fails because that address doesn't exist.

 Any other ideas?

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Anthony
 Rodgers
 Sent: Thursday, July 13, 2006 2:24 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Email notification of voicemail

 Try having nothing after the name in your voicemail.conf:

 1234 = 1234,The Marquis de Sade

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RE: [BULK] Re: [Asterisk-Users] how to decrease answer time !

2006-07-13 Thread Kevin Savoy

The world is full of smart alecks. Thank the lords because what a boring
world this would be without us :)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer
Sent: Thursday, July 13, 2006 4:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [BULK] Re: [Asterisk-Users] how to decrease answer time !
Importance: Low

When you hear the phone ring, run faster so you reach it more quickly.

On 13/07/06, Pablo Mora [EMAIL PROTECTED] wrote:







 Pablo Mora, Ing.

 GERENTE DE OPERACIONES

 ESPOLTEL S.A.

 Malecón 100 y Loja

 Telf.:2514477


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-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
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[asterisk-users] Email notification of voicemail

2006-07-12 Thread Kevin Savoy









Asterisk
is trying to send an email to users when they receive a voicemail. Can this be
shut off? I have not entered any email addresses in voicemail.conf so it tries
to send to [EMAIL PROTECTED].
This of course gets rejected since the user does not exist and the root users
mailbox on linux gets full of these rejection notices. I cant seem to
find anywhere to tell Asterisk to stop notifying people they have voicemails.



Im
using 1.2.9.1 of Asterisk. Thanks



_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [asterisk-users] Email notification of voicemail

2006-07-12 Thread Kevin Savoy

I have attach=no in my voicemail.conf so that can't be doing it. Not sure
where that sendmail command is. Don't see it in voicemail.conf or any other
config in the asterisk directory.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VoIP Street
Sent: Wednesday, July 12, 2006 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Email notification of voicemail

Kevin Savoy wrote:
 Asterisk is trying to send an email to users when they receive a 
 voicemail. Can this be shut off? I have not entered any email addresses 
 in voicemail.conf so it tries to send to [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]. This of course gets rejected 
 since the user does not exist and the root users mailbox on linux gets 
 full of these rejection notices. I can't seem to find anywhere to tell 
 Asterisk to stop notifying people they have voicemails.

 I'm using 1.2.9.1 of Asterisk. Thanks

 _

 **Kevin Savoy**

 **Business Unit Telecom Analyst**

 2218 4th Ave W

 Williston, ND 58801

 Ph: 701-774-4023

 Fax: 701-774-2901

 http://www.novo1.com

 Novo 1 is a service mark of Novo 1, Inc

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You could try commenting out:

attach=yes

Also, if you don't want any emails sent ever for any voice mail users 
you could probably uncomment the following line and give it a bogus path 
to the mailer.

;mailcmd=/usr/sbin/sendmail -t

There is probably a better way to do this but we have never needed to 
turn it off so I am not sure.

Hope this helps.

-- 
VoIP Street
Origination/Termination with SUPERIOR customer service!
http://www.VoIPstreet.com
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RE: [asterisk-users] Unable to configure my DID number

2006-07-10 Thread Kevin Savoy









Are you sure they are sending you all 10
digits and not just the last four? Our provider just sends the last four digits
on DID. If this is the case you would have this:



exten = 4567,1,Answer()
exten = 4567,1,DIAL(SIP/user,20)



Hope this helps.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Monday, July 10, 2006 7:45
AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] Unable
to configure my DID number





Hi friends,

At present, I am making outgoing calls using Teliax service with Asterisk. But,
I am unable to receive calls. My DID number is: 3031234567. I am using SIP
Server (Asterisk) setup, which is provided on Teliax website support. I have
replaced my DID number i.e., 3031234567 in YOURNUMBER. But, I am unable to
receive calls. 

My configuration file in extensions.conf File:

exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)

exten = 3031234567,1,Answer()
exten = 3031234567,1,DIAL(SIP/user,20)

---
I hope the above configuration is proper, If not please suggest the
modifications.

In addition, I have some doubts.

1) How should I configure my DID number in extensions.conf file to recevice
incoming calls?
2) Are they any modifications required in Features option in my
account on Teliax website?
3) To receive incoming calls, do I need to make any kind of modifications to
other configuration files in Asterisk and setup DID number?
4) Do I need to set Public IP in my Asterisk server or our local IP is enough?
4) After configuring DID number, where can I receive the phone call (ring)?
5) How can I setup IVR (Interactive Voice Response) system to my DID number.
(i.e., If someone calls to my DID number, then our IVR (Welcome message) should
respond and ask for extension number.)

Please respond to this message ASAP. Looking forward to your response.

Thank you.

Regards,
Chandra.

 







How low will we go? Check out Yahoo! Messengers low PC-to-Phone
call rates.






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RE: [asterisk-users] RE: Asterisk in Seattle

2006-07-06 Thread Kevin Savoy

I'm in Williston, North Dakota and we have an office in Billings, MT. He's
right. We are 500 miles form civilization! :)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, July 05, 2006 10:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] RE: Asterisk in Seattle

It can't be that bad there in Anchorage. I'm in Billings, MT, which is about
half the size of Anchorage, and sometimes (no, wait... most times) it seems
like I'm 500 miles from civilsation. Wait, I AM 500 miles from civilisation!

-Original Message- 
From: Josh Reineke [mailto:[EMAIL PROTECTED] 
Sent: Wed 7/5/2006 8:03 PM 
To: asterisk-users@lists.digium.com 
Cc: 
Subject: [asterisk-users] RE: Asterisk in Seattle

I work for a medium size business in Anchorage, AK running two
installations with about 30 handsets a piece.  They've both been in
service for a couple of years.

I'm in Seattle fairly frequently, being it's the metropolis closest
to
Anchorage.  I'd be jazzed if there was a user group there and would
be
willing to help in it's formation.

Josh

Message: 15
Date: Wed, 5 Jul 2006 14:00:35 -0600
From: Douglas Garstang [EMAIL PROTECTED]
Subject: [asterisk-users] Asterisk in Seattle
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:

[EMAIL PROTECTED]
Content-Type: text/plain;   charset=iso-8859-1

All,

Anyone know of any companies (small, large) that are using,
experimenting with, deploying, and so on, Asterisk in Washington
state,
most likely in and around Seattle? I'm curious from an employment
perspective. :)

Doug.
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[asterisk-users] Zap Channel not hanging up on Telco side

2006-07-06 Thread Kevin Savoy









Im
having an issue where Asterisk hangs up a call (either party hangs up) but the
telco side of the T1, both the local company and ATT, does not receive the
hangup signal from Asterisk. Therefore Asterisk thinks the channel is available
but its still off-hook on the telco side. I have confirmed this with
ATT that the channels are open on their side but not in Asterisk. Any
ideas what is causing this? The CLI does show the call hanging up. This is not
on every call and Ive not been able to determine yet when it locks and
when it doesnt. My guess at this point is when a call is placed but the
caller does not wait for the answering party to answer and hangs up. Then the
called party line keeps ringing, Ive confirmed this calling my cell phone,
and then the called party answers. At this point I see the CLI state that it is
starting a simple switch, as if the call is reconnecting to Asterisk and then
hangs up since I have the s,1,Hangup() and Asterisk at this point doesnt
know where the call should go. I believe at this point the call is hungup in
Asterisk but NOT on the telco side.



Any
ideas how to get around this???



Thanks



_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [asterisk-users] Zap Channel not hanging up on Telco side

2006-07-06 Thread Kevin Savoy

Good point. I am using Asterisk 1.2.9.1, Zaptel 1.2.6. libpri 1.2.3. I had
the same issue with Asterisk 1.2.7.1, Zaptel 1.2.5 and libpri 1.2.2.

My Zapata.conf looks like this:

[channels]
context=default
musiconhold=default
resetinterval=60

;ATT T1's
group=1
switchtype=national
signalling=em_w
channel = 1-96

;Local T1's
group=2
switchtype=national
signalling=em_w
channel = 97-168

My Zaptel.conf looks like this. We are using RedFone's Fonebridges hence the
dynamic channels but I was able to re-create this with the wct4xxp card we
have as well.

#
#ATT
#
dynamic=eth,eth0/00:0C:42:03:63:0F/0,24,1
em=1-24
dynamic=eth,eth0/00:0C:42:03:63:0F/1,24,2
em=25-48
dynamic=eth,eth0/00:0C:42:03:63:0F/2,24,3
em=49-72
dynamic=eth,eth0/00:0C:42:03:63:0F/3,24,4
em=73-96
#
#Local
#
dynamic=eth,eth0/00:0C:42:03:63:17/0,24,1
em=97-120
dynamic=eth,eth0/00:0C:42:03:63:17/1,24,2
em=121-144
dynamic=eth,eth0/00:0C:42:03:63:17/2,24,3
em=145-168
#dynamic=eth,eth0/00:0C:42:03:63:17/3,24,4
#em=169-192

loadzone=us
defaultzone=us


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, July 06, 2006 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Channel not hanging up on Telco side

Kevin Savoy wrote:

 I'm having an issue where Asterisk hangs up a call (either party hangs 
 up) but the telco side of the T1, both the local company and ATT, 
 does not receive the hangup signal from Asterisk. Therefore Asterisk 
 thinks the channel is available but it's still off-hook on the telco


Does us no good, not knowing what version of Asterisk, or seeing how you 
have it configured. How about showing your zaptel, zapata and relevant 
configs?

Doug

-- Ben Franklin quote: Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety.

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RE: [asterisk-users] Zap Channel not hanging up on Telco side

2006-07-06 Thread Kevin Savoy

No I have not using bristuff.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of stoffell
Sent: Thursday, July 06, 2006 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Channel not hanging up on Telco side

On 7/6/06, Kevin Savoy [EMAIL PROTECTED] wrote:
 I'm having an issue where Asterisk hangs up a call (either party hangs up)
but the telco side
 of the T1, both the local company and ATT, does not receive the hangup
signal from
 Asterisk. Therefore Asterisk thinks the channel is available but it's
still off-hook on the telco

I have not experienced this on a standard asterisk yet, but I did on a
bristuff version (the latest 0.3.0pre-1 series, being n, o, p and q).
However, there's a patch to libpri for this in the mailing list.

Are you using the bristuff'ed version?

cheers
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[asterisk-users] Possible Bug?

2006-07-05 Thread Kevin Savoy









I
believe Ive found a possible bug in the Zaptel channel drivers. Ive
been able to recreate this on a couple of servers. One with Asterisk 1.2.7.1
and 1.2.9.1,and also Zaptel 1.2.5 and Zaptel 1.2.6. I was testing a server
configuration with some T1s we have with ATT. When I disconnected
the ATT T1s we were no longer able to check voicemail. I noticed
then that no messages from the server could be heard. Playback, voicemail or
any other message the server would play to the caller on the phone. It would
display on the CLI that it was playing the message but nothing could be heard
and it would hang the call until it timed out. It took a lot of trial and error
before I figured out that if I unloaded all related files to the Zaptel drivers
the messages could then be heard. Reactivated the Zaptel with no T1s
attached and again it killed the messages. One server has a Wct4xxp card and
the other using dynamic with Redfones from Fonbridge. Both reacted the
same way.



This
would not be good if we had a fiber cut somewhere and then our users could no
longer use their voicemail or any other application that played sounds.



Can
anyone else recreate this? Do any of you think this is a bug? Should I submit a
bug report?



Ive
heard to submit a bug to send it to mantis. What is this and where or how do I
do this?



Thanks



_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [Asterisk-Users] bristuff hangup issue

2006-06-29 Thread Kevin Savoy

I can also add that this happens on em_w lines as well. I've had issues
where callers start getting dead air when dialing out. Talking with the
phone company the lines were in an off-hook state even though Asterisk hung
up the call. I done exactly as below where I hang up before the other party
answers the call. I've also had where after I hang up the CLI shows the call
hanging up and then another call starts, starting simple switch, as if the
call was re-established but Asterisk doesn't know what to do with the call
and executes the s,1,hangup() on the call. This does NOT however always hang
up the call on the ATT side. The T1 still shows the call as off-hook even
though it's not in use. It seems random (at least I haven't figured out the
pattern yet) as to when the channel gets hung up properly on ATT's side and
when it's not.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of stoffell
Sent: Thursday, June 29, 2006 3:44 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] bristuff hangup issue

hi,

Just wanted to inform everyone, if you're using the latest bristuff's
you might (depends on the country!) have hangup issues.

The issue appears every time you dial an external number, and hangup
after letting it ring for a few times. Then the remote party keeps
ringing.

In some situations (we only encountered this while dialing to other *
servers) it keeps the line open on the telco-side. Meaning.. you pay
for it! The cdr on the calling asterisk (with the bug) doesn't
indicate a long connection time. However, the cdr on the called
asterisk does.. (I've seen several durations of over 20 hours) A show
channels doesn't indicate any active calls.

A quick fix has been posted a while ago by Marcel van der Boom (in
libpri/q931.c), this works. According to the release notes this should
have been applied to the latest bristuff, but be careful, the problem
still exists on bristuff-0.3.0-PRE-1q.

I have emailed junghanns.net to let them know.

Best regards,

stoffell
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RE: [Asterisk-Users] Dropping incompatible voice frame

2006-06-29 Thread Kevin Savoy









This didnt work for me either. I
tried using the patch at the link below and it didnt work either. 



If I were to guess what was happening
here, it would be when the call is forwarded by the phone Asterisk doesnt
know which device to send the call to. How does it know to open a Zap channel
and dial the command? What tells Asterisk to open Zap channel and dial the
number the phone had it its forward? Am I off track here?











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail
Sent: Wednesday, June 28, 2006
10:24 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Dropping incompatible voice frame







This is known issue, we fixed it by putting an answer() in the
dial plan before it gets forwarded,thefix
transcode_via_sln=no (detailed in the bug tracker)didn't work for me.
YMMV.












http://bugs.digium.com/view.php?id=4101






On 6/28/06, Kevin
Savoy [EMAIL PROTECTED]
wrote: 







Sorry
if this has been posted before but I'm having an issue where I get the
following on my CLI.



ast_read:
Dropping incompatible voice frame on Local/XX of format ulaw since our native form has changed to slin



A
call comes in on our main to toll free number on an ATT T1 line and is
sent to phone 4000. This is our secretary's desk. If she leaves the desk she
forwards the phone to one of our sister companies so that they would answer the
call. This call is sent back out the ATT T1. If she answers the call and
then forwards outside the building it works fine but if she forwards her phone
outside the building to auto forward the call when she is away from her desk we
get the above error. I have recreated this on my own phone (both hers and mine
are Polycom 501's) and with a Cisco 7960. I also tried a different toll free
number with the same results. I searched the internet and found four people
having the same issue but none have gotten responses on how to fix it. Each
time it was something similar where the call was redirected. I know the T1's
are configured correctly because all other incoming and outgoing calls work
fine until this error occurs. Then nothing works. 



I
am using Asterisk 1.2.7.1 with
Zaptel 1.2.5 and Libpri 1.2.2 . I have tried using both a digium Wctxxp 4 port
and RedFone's Fonebridges and have gotten the same result both ways so the
problem is within Asterisk itself. I also tried allow=all in sip.conf as
well as specifically listing allow= slin
and all other formats to no avail. 



Also
when this happens the channel is no longer usable even though Asterisk thinks
it is available. When the next call is placed it times out because that channel
has been locked by the above error. The only way out is a complete reboot and
reset of all systems. Not good. 



Any
help would be greatly appreciated. If I had hair left I'd be pulling it out
about now.





Thanks

_



Kevin Savoy

Business Unit Telecom Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc










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[Asterisk-Users] Dropping incompatible voice frame

2006-06-28 Thread Kevin Savoy









Sorry
if this has been posted before but Im having an issue where I get the following
on my CLI.



ast_read:
Dropping incompatible voice frame on Local/XX of format ulaw since our
native form has changed to slin



A
call comes in on our main to toll free number on an ATT T1 line and is sent
to phone 4000. This is our secretarys desk. If she leaves the desk she forwards
the phone to one of our sister companies so that they would answer the call. This
call is sent back out the ATT T1. If she answers the call and then forwards
outside the building it works fine but if she forwards her phone outside the building
to auto forward the call when she is away from her desk we get the above error.
I have recreated this on my own phone (both hers and mine are Polycom 501s)
and with a Cisco 7960. I also tried a different toll free number with the same
results. I searched the internet and found four people having the same issue
but none have gotten responses on how to fix it. Each time it was something
similar where the call was redirected. I know the T1s are configured
correctly because all other incoming and outgoing calls work fine until this
error occurs. Then nothing works.



I am
using Asterisk 1.2.7.1 with Zaptel 1.2.5 and Libpri 1.2.2. I have tried using
both a digium Wctxxp 4 port and RedFones Fonebridges and have gotten the
same result both ways so the problem is within Asterisk itself. I also tried
allow=all in sip.conf as well as specifically listing allow=slin and all other
formats to no avail. 



Also
when this happens the channel is no longer usable even though Asterisk thinks
it is available. When the next call is placed it times out because that channel
has been locked by the above error. The only way out is a complete reboot and
reset of all systems. Not good. 



Any
help would be greatly appreciated. If I had hair left Id be pulling it
out about now.





Thanks

_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [Asterisk-Users] Turning AAAH into a call-center

2006-05-16 Thread Kevin Savoy

Speaking as one of those call centers we are looking at doing a turn over to
Asterisk from our Nortel systems and are doing it ourselves. We've looked at
a lot of packages from Fonality, Signate, Aheeva and others and none fit our
needs. Each has good aspects but none have all of what we need. Below he
states extensive reporting in QueueMetrics. Yes and no. Extensive but
limited to a few areas. Agent level reporting doesn't go deep enough. There
is no DNIS reporting. In our business that is critical. We have 40 to 50
queues and some queues can have up to 1,000 toll free numbers pointed at it.
Our clients want to know how many calls on each individual toll free and all
the statistics that go with it. Talk time, abandons, thresholds, etc.

The commercial products are fine products but not in depth enough for a
larger call center. We have decided to look into doing it ourselves and
found that it wasn't really that difficult to find all the numbers we
needed. It left me feeling that these companies didn't really spend a lot of
time on reporting as we found all we needed pretty quickly. The previously
stated companies do fine with small call centers or call centers with only a
few clients.

We're having more issues with finding a phone to fit our business then we
are software for Asterisk. I'm disappointed that all these phone
manufacturers have never considered the call center when designing phones.
Nortel and Avaya have phones specifically designed for call centers. They
don't have handsets. Only headsets. Sadly none of the VoIP phones are
designed that way. 

Anyways that's my 2 cents on this. (if anyone cares) :)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent: Tuesday, May 16, 2006 6:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Turning AAAH into a call-center


I believe there are quite different levels for the Asterisk market, so  
most people who run call centers wont feel confident in downloading a  
couple of ISOs from the internet and setting things up themselves.
l.


In data Mon, 15 May 2006 21:47:09 +0200, Steve Totaro  
[EMAIL PROTECTED] ha scritto:

 I bet Signate will love this.

 Lenz wrote:
 Hello list,
 we have prepared a short tutorial that will teach you to turn your  
 [EMAIL PROTECTED] box into a full-fledged call center within minutes, with  
 both always-on and callback agents available and the very extensive  
 reporting facilities that QueueMetrics provides.

 You can download it from the donwloads page at  
 http://queuemetrics.loway.it/download.jsp

 QueueMetrics is a full-fledged call-center monitoring system and it is  
 available for free to home users, SOHOs and individual enthusiasts.

 Any comment on the document is welcome.
 l.


 --Loway Research - Home of QueueMetrics
 http://queuemetrics.loway.it

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-- 
Assum est, versa et manduca.
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-09 Thread Kevin Savoy









I see two problems with doing it this way.
One the agent is not paying attention and misses the announcement. Two we have
some queues that can have as many 800+ toll free numbers pointed at the same
queue. The agents need to be able to know which of the toll free numbers was
dialed. We do that in our current system by putting the last four digits of the
toll free on the phone display. With the below scenario it will not only have
to have a variable as to which account but also say out the digits of the last
four but only on certain accounts. That is possible I realize but messy. 



I need a combination of the two scenarios.
AgentCallBackLogin so that I can have a phone display but AgentLogin so that
there is no Call Back and just presents the call. Unless I can come up with
something Im guessing we are going to have to write some kind of program
that can do this instead.



Anyone have anything else to add? Thanks











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Junaid Uppal
Sent: Tuesday, May 09, 2006 2:57
AM
To: Asterisk Users Mailing List -
 Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer





Hi Steve ,

I was actually looking forward for the same thing , do y ou have something like
this , as an example?

regards

Junaid Uppal





On 5/9/06, Steve
Totaro [EMAIL PROTECTED]
wrote:

Use an activex screenpop.

Thanks,
Steve Totaro

 -Original Message-
 From: Kevin Savoy [mailto:[EMAIL PROTECTED]]
 Sent: Monday, May 08, 2006 3:32 PM 
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer

 This may be the way to go but not the best. Our agents frankly aren't 
the
 brightest people and I can see them forgetting it as soon as it is
said to
 them, or they are not paying attention and missing the announcement
but it
 is something to look into. Thanks 

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:
[EMAIL PROTECTED]] On Behalf Of Time
Bandit
 Sent: Monday, May 08, 2006 2:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer 

  Ok I can get this to work now the next problem is since the agent
stays
  off-hook when a call is presented to them there is no
indication
of
 what
  call this is. Being an inbound call center we have 100's of clients. 
 1,000's
  of toll frees and DNIS. We use the Asterisk callerID function to
assign
 a
  name to each call so that when the call is presented to the agent it
  displays which company the call is for. With AgentLogin all the 
agent
 gets
  is the number they dialed to log in. No idea which client this call
is
 for.
  Any ideas there?
 When you send the caller to the queue, you can pass the name of the 
 audio file to be played as the announcement to the agent when he gets
 the call. Maybe you could use that and pre-record the name of the
 customer, passing that audio file

 something like exten = 
 8000,n,Queue(8000|t||FilenameOfTheCustomerName|300)

 maybe you could also use festival

 hth
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-09 Thread Kevin Savoy

This is an idea I've had for long term. Might have to push it up a bit. Not
with astGUI but a similar CTI idea. Using the inbound DNIS to trigger screen
pops.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: Tuesday, May 09, 2006 6:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Hello,

It may be more than you want, but astGUIclient/VICIDIAL has screen
pops for inbound calls through a web browser. It is also Open-source.
http://astguiclient.sf.net

MATT---

On 5/9/06, Steve Totaro [EMAIL PROTECTED] wrote:
 I do but I didn't write it and it was on company time so I cannot
 share.  Sorry.  Just throwing out ideas and possibilities.

 Junaid Uppal wrote:
  Hi Steve ,

  I was actually looking forward for the same thing , do y ou have
  something like this , as an example?

  regards

  Junaid Uppal

  On 5/9/06, * Steve Totaro* [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED] wrote:

  Use an activex screenpop.

  Thanks,
  Steve Totaro

   -Original Message-
   From: Kevin Savoy [mailto:[EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED]]
   Sent: Monday, May 08, 2006 3:32 PM
   To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer

   This may be the way to go but not the best. Our agents frankly
  aren't
  the
   brightest people and I can see them forgetting it as soon as it is
  said to
   them, or they are not paying attention and missing the
announcement
  but it
   is something to look into. Thanks

   -Original Message-
   From: [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED]
   [mailto: [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED]] On Behalf Of Time
  Bandit
   Sent: Monday, May 08, 2006 2:23 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Ok I can get this to work now the next problem is since the
agent
  stays
off-hook when a call is presented to them there is no
indication
  of
   what
call this is. Being an inbound call center we have 100's of
  clients.
   1,000's
of toll frees and DNIS. We use the Asterisk callerID function to
  assign
   a
name to each call so that when the call is presented to the
  agent it
displays which company the call is for. With AgentLogin all the
  agent
   gets
is the number they dialed to log in. No idea which client this
  call
  is
   for.
Any ideas there?
   When you send the caller to the queue, you can pass the name of
the
   audio file to be played as the announcement to the agent when he
  gets
   the call. Maybe you could use that and pre-record the name of the
   customer, passing that audio file

   something like exten =
   8000,n,Queue(8000|t||FilenameOfTheCustomerName|300)

   maybe you could also use festival

   hth
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-08 Thread Kevin Savoy









Firstly the auto-answer on both the 301
and 501 phone is set to on, but it doesnt seem to have an effect. Ill
have to look into this _ALERT_INFO variable. Not much experience with it here. 

Could you give me a dial plan example that
would work? Here is what we have now.



exten=3472,1,Answer()

exten=3472,2,Wait(1)

exten=3472,3,Playback(this-call-may-be-monitored-or-recorded)

exten=3472,4,SetCallerID(ICS)

exten=3472,5,Queue(ICS)



What can I add to this to make the phone
auto-answer? Thanks











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer
Sent: Friday, May 05, 2006 6:29 PM
To:
asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer







I don't see any reason you can't use a polycom. You should be able to
solve your problem multiple ways. You can simply put the default ring on the
Polycom to autoanswer if that is the sole purpose, give it a second extension
to be used in the queue that is programmed to autoanswer, as a couple of
examples, or design your dialplan such that the appropriate _ALERT_INFO
variable is set where the queue is concerned.











p












From: Kevin Savoy [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Date: Fri, 5 May 2006 15:31:41 -0500
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer

The problem with what is in wiki is that these calls are being sent to a
queue. There is no way to have the queue dial the preceding digit that I can
think of that would trigger this. In the example shown he has an 8 dialed
before the extension. How would I get Asterisk to dial an 8 before sending
the call to the logged in agent in the queue?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark
Sent: Friday, May 05, 2006 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote:

 Can anyone recommend a phone to use in an inbound call center 
 environment that has an auto answer feature? We don't want the agents 
 having to acknowledge the call. The call should just activate on the 
 headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
 None of these support it.

My Polycom phones support auto-answer. This link should get you started.

http://www.voip-info.org/wiki-Polycom+auto-answer+config

Mike
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-08 Thread Kevin Savoy









Correct. We have to hit the answer
button. In a call center environment such as ours we dont want to give
the agents the option of not answering the call when they are logged in.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Monday, May 08, 2006 10:15
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





Does the phone ring, just not auto-answer?





Thanks,
Steve Totaro

 













From: Kevin Savoy
[mailto:[EMAIL PROTECTED] 
Sent: Monday, May 08, 2006 10:17
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





Firstly the auto-answer on both the 301
and 501 phone is set to on, but it doesnt seem to have an effect.
Ill have to look into this _ALERT_INFO variable. Not much experience
with it here. 

Could you give me a dial plan example that
would work? Here is what we have now.



exten=3472,1,Answer()

exten=3472,2,Wait(1)

exten=3472,3,Playback(this-call-may-be-monitored-or-recorded)

exten=3472,4,SetCallerID(ICS)

exten=3472,5,Queue(ICS)



What can I add to this to make the phone
auto-answer? Thanks











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer
Sent: Friday, May 05, 2006 6:29 PM
To:
asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer







I don't see any reason you can't use a polycom. You should be able to
solve your problem multiple ways. You can simply put the default ring on the
Polycom to autoanswer if that is the sole purpose, give it a second extension
to be used in the queue that is programmed to autoanswer, as a couple of
examples, or design your dialplan such that the appropriate _ALERT_INFO
variable is set where the queue is concerned.











p












From: Kevin Savoy [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Date: Fri, 5 May 2006 15:31:41 -0500
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer

The problem with what is in wiki is that these calls are being sent to a
queue. There is no way to have the queue dial the preceding digit that I can
think of that would trigger this. In the example shown he has an 8 dialed
before the extension. How would I get Asterisk to dial an 8 before sending
the call to the logged in agent in the queue?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark
Sent: Friday, May 05, 2006 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote:

 Can anyone recommend a phone to use in an inbound call center 
 environment that has an auto answer feature? We don't want the agents 
 having to acknowledge the call. The call should just activate on the 
 headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
 None of these support it.

My Polycom phones support auto-answer. This link should get you started.

http://www.voip-info.org/wiki-Polycom+auto-answer+config

Mike
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-08 Thread Kevin Savoy









Because we will have many of these phones
in remote locations and we dont want to be chewing up bandwidth with
agents not on calls. Am I making the right assumption here that phones that are
idle will not be taking up bandwidth where ones with MOH playing would be?











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Monday, May 08, 2006 10:40
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer







Why not just use AgentLogin and let them
listen to music until a call comes in?



Thanks,
Steve Totaro

 













From: Kevin Savoy
[mailto:[EMAIL PROTECTED] 
Sent: Monday, May 08, 2006 11:27
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





Correct. We have to hit the
answer button. In a call center environment such as ours we
dont want to give the agents the option of not answering the call when they
are logged in.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Monday, May 08, 2006 10:15
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





Does the phone ring, just not auto-answer?





Thanks,
Steve Totaro

 













From: Kevin Savoy
[mailto:[EMAIL PROTECTED] 
Sent: Monday, May 08, 2006 10:17
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





Firstly the auto-answer on both the 301
and 501 phone is set to on, but it doesnt seem to have an effect.
Ill have to look into this _ALERT_INFO variable. Not much experience
with it here. 

Could you give me a dial plan example that
would work? Here is what we have now.



exten=3472,1,Answer()

exten=3472,2,Wait(1)

exten=3472,3,Playback(this-call-may-be-monitored-or-recorded)

exten=3472,4,SetCallerID(ICS)

exten=3472,5,Queue(ICS)



What can I add to this to make the phone
auto-answer? Thanks











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer
Sent: Friday, May 05, 2006 6:29 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer







I don't see any reason you can't use a polycom. You should be able to
solve your problem multiple ways. You can simply put the default ring on the
Polycom to autoanswer if that is the sole purpose, give it a second extension
to be used in the queue that is programmed to autoanswer, as a couple of
examples, or design your dialplan such that the appropriate _ALERT_INFO
variable is set where the queue is concerned.











p












From: Kevin Savoy [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Date: Fri, 5 May 2006 15:31:41 -0500
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer

The problem with what is in wiki is that these calls are being sent to a
queue. There is no way to have the queue dial the preceding digit that I can
think of that would trigger this. In the example shown he has an 8 dialed
before the extension. How would I get Asterisk to dial an 8 before sending
the call to the logged in agent in the queue?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark
Sent: Friday, May 05, 2006 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote:

 Can anyone recommend a phone to use in an inbound call center 
 environment that has an auto answer feature? We don't want the agents 
 having to acknowledge the call. The call should just activate on the 
 headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
 None of these support it.

My Polycom phones support auto-answer. This link should get you started.

http://www.voip-info.org/wiki-Polycom+auto-answer+config

Mike
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FareChase 










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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-08 Thread Kevin Savoy









Ok I can get this to work now the next
problem is since the agent stays off-hook when a call is
presented to them there is no indication of what call this is. Being an inbound
call center we have 100s of clients. 1,000s of toll frees and
DNIS. We use the Asterisk callerID function to assign a name to each call so
that when the call is presented to the agent it displays which company the call
is for. With AgentLogin all the agent gets is the number they dialed to log in.
No idea which client this call is for. Any ideas there?











From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Steve Totaro
Sent: Monday, May 08, 2006 11:23
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





That is correct. Just use IAX
trunking and speex. You will be fine.





Thanks,
Steve Totaro

 













From: Kevin Savoy
[mailto:[EMAIL PROTECTED] 
Sent: Monday, May 08, 2006 12:12 PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





Because we will have many of these phones
in remote locations and we dont want to be chewing up bandwidth with
agents not on calls. Am I making the right assumption here that phones that are
idle will not be taking up bandwidth where ones with MOH playing would be?











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Monday, May 08, 2006 10:40
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer







Why not just use AgentLogin and let them
listen to music until a call comes in?



Thanks,
Steve Totaro

 













From: Kevin Savoy
[mailto:[EMAIL PROTECTED] 
Sent: Monday, May 08, 2006 11:27
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





Correct. We have to hit the
answer button. In a call center environment such as ours we
dont want to give the agents the option of not answering the call when
they are logged in.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Monday, May 08, 2006 10:15
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





Does the phone ring, just not auto-answer?





Thanks,
Steve Totaro

 













From: Kevin Savoy
[mailto:[EMAIL PROTECTED] 
Sent: Monday, May 08, 2006 10:17
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer





Firstly the auto-answer on both the 301
and 501 phone is set to on, but it doesnt seem to have an effect.
Ill have to look into this _ALERT_INFO variable. Not much experience
with it here. 

Could you give me a dial plan example that
would work? Here is what we have now.



exten=3472,1,Answer()

exten=3472,2,Wait(1)

exten=3472,3,Playback(this-call-may-be-monitored-or-recorded)

exten=3472,4,SetCallerID(ICS)

exten=3472,5,Queue(ICS)



What can I add to this to make the phone
auto-answer? Thanks











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer
Sent: Friday, May 05, 2006 6:29 PM
To:
asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer







I don't see any reason you can't use a polycom. You should be able to
solve your problem multiple ways. You can simply put the default ring on the
Polycom to autoanswer if that is the sole purpose, give it a second extension
to be used in the queue that is programmed to autoanswer, as a couple of
examples, or design your dialplan such that the appropriate _ALERT_INFO
variable is set where the queue is concerned.











p












From: Kevin Savoy [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Date: Fri, 5 May 2006 15:31:41 -0500
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer

The problem with what is in wiki is that these calls are being sent to a
queue. There is no way to have the queue dial the preceding digit that I can
think of that would trigger this. In the example shown he has an 8 dialed
before the extension. How would I get Asterisk to dial an 8 before sending
the call to the logged in agent in the queue?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark
Sent: Friday, May 05, 2006 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote:

 Can anyone recommend a phone to use in an inbound call center 
 environment that has an auto answer feature? We don't want the agents 
 having to acknowledge the call. The call should just activate on the 
 headphones. We have tried Grandstream 2000

RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-08 Thread Kevin Savoy

This may be the way to go but not the best. Our agents frankly aren't the
brightest people and I can see them forgetting it as soon as it is said to
them, or they are not paying attention and missing the announcement but it
is something to look into. Thanks 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Monday, May 08, 2006 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

 Ok I can get this to work now the next problem is since the agent stays
 off-hook when a call is presented to them there is no indication of what
 call this is. Being an inbound call center we have 100's of clients.
1,000's
 of toll frees and DNIS. We use the Asterisk callerID function to assign a
 name to each call so that when the call is presented to the agent it
 displays which company the call is for. With AgentLogin all the agent gets
 is the number they dialed to log in. No idea which client this call is
for.
 Any ideas there?
When you send the caller to the queue, you can pass the name of the
audio file to be played as the announcement to the agent when he gets
the call. Maybe you could use that and pre-record the name of the
customer, passing that audio file

something like exten =
8000,n,Queue(8000|t||FilenameOfTheCustomerName|300)

maybe you could also use festival

hth
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[Asterisk-Users] Dumping queue_log to MySQL

2006-05-05 Thread Kevin Savoy









Anyone
have a working solution for this? I played with the demo that came with
QueueMetrics to see how they were doing it and it was working for a bit but now
somehow every night it stopped. Perl and Tail are still running on the server
but the information is not dumping to the MySQL database. I dont get any
error messages anywhere telling me why it stops. As far as tail and perl are concerned
everything is fine. 

We
will be using this for a call center and need more reliability. Anyone got one
working?



Thanks



_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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[Asterisk-Users] Call Center Phone with Auto Answer

2006-05-05 Thread Kevin Savoy









Can anyone
recommend a phone to use in an inbound call center environment that has an auto
answer feature? We dont want the agents having to acknowledge the call. The
call should just activate on the headphones. We have tried Grandstream 2000,
Polycom 301, 501 and 601. None of these support it.



Thanks



_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-05 Thread Kevin Savoy

The problem with what is in wiki is that these calls are being sent to a
queue. There is no way to have the queue dial the preceding digit that I can
think of that would trigger this. In the example shown he has an 8 dialed
before the extension. How would I get Asterisk to dial an 8 before sending
the call to the logged in agent in the queue?

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark
Sent: Friday, May 05, 2006 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote:

 Can anyone recommend a phone to use in an inbound call center 
 environment that has an auto answer feature? We don't want the agents 
 having to acknowledge the call. The call should just activate on the 
 headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
 None of these support it.

My Polycom phones support auto-answer. This link should get you started.

http://www.voip-info.org/wiki-Polycom+auto-answer+config

Mike
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

2006-05-05 Thread Kevin Savoy

We are using agent login but we don't want MOH on the line at all times as
some of these phones could and probably will be connected in remote
locations. We don't want to stream MOH across frames chewing up bandwidth
when there are no calls to present to that phone. We do have the ackcall=no
in the agents.conf and it seems to have no affect.

Am I missing something here? Appreciate any help

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Cook
Sent: Friday, May 05, 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote:

 Can anyone recommend a phone to use in an inbound call center 
 environment that has an auto answer feature? We don't want the agents 
 having to acknowledge the call. The call should just activate on the 
 headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
 None of these support it.

 Thanks

May be I am not understanding... Why not use agentlogin and have the 
agents always logged in with MOH... they get a beep and they are 
connected.. Change ackcall=no in agents.conf

Then you don't need auto-answer
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RE: [Asterisk-Users] Auto Logout from queue

2006-05-04 Thread Kevin Savoy

I have tried using the autologoff in the agents.conf and it sort of works. I
set it to 5 seconds to test it and it has taken anywhere from 35 to 60
seconds to actually do something at which point it does indeed log out the
agent.

I don't want to be pestering agents with test calls to see if they are
indeed there so the below scripting isn't really practical in our
environment. 

Can anyone tell me why the agents.conf file setting doesn't work as
described? If it is set to 5 it should log them off after 5 seconds or so
not 30 - 60 seconds. I don't really want the call sitting at a logged out
agents phone for anymore then 5 seconds when there are other agents out
there waiting to take that call. Any ideas? 

Thanks

_
 
Kevin Savoy
Business Unit Telecom Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http://www.novo1.com
Novo 1 is a service mark of Novo 1, Inc

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
Sagredo
Sent: Tuesday, April 25, 2006 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Auto Logout from queue

Via dialplan maybe?

exten = xxx,1,Dial(SIP/101_Queue,20,tr)
exten =xxx,2,RemoveQueueMember(Comercial_Queue,SIP/101_Queue,1)



Kerry Garrison escribió:
 Yes, that is the functionality I am looking for, just not sure how exactly
 to pull that off.


   _  

 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Alexander
 Lopez
 Sent: Tuesday, April 25, 2006 12:08 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Auto Logout from queue


 Use the local channel to call the agent first, and if there is no answer,
 log them out.
  
  

   _  

 From: [EMAIL PROTECTED] on behalf of Kerry Garrison
 Sent: Tue 4/25/2006 2:38 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] Auto Logout from queue


 i have a client that wants a function that will automatically logout an
 agent from a queue if they do not answer a call. This would prevent future
 calls from being sent to that phone if the agent forgot to logout. Any
 ideas?
  
 Kerry Garrison
 Director of Technical Services
 Tech Data Pros - Orange County's Mobile IT Service Provider
 (949) 502-7819 x200 -  mailto:[EMAIL PROTECTED]
 [EMAIL PROTECTED]
  http://www.techdatapros.com/ http://www.techdatapros.com 
  

   
 

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[Asterisk-Users] Status of Queue

2006-04-26 Thread Kevin Savoy









Is
there variable or a way I can check to ensure that an agent is logged into a
particular queue? I dont want to queue a call up to a queue that no one
is logged into. I would like to have the call redirected to another extension
if there are no queue members.



Thanks
for any insight



_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [Asterisk-Users] PRI blocking on incoming calls

2006-04-19 Thread Kevin Savoy









We tried the pri intese debug last night but this only
showed us what was going on with the Nortel trunk not the MCI T1 side. Sadly it
was of no use. Any other ideas?



Anyone??



Thanks











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oscar Carriles
Sent: Tuesday, April 18, 2006 2:48
PM
To: 'Asterisk Users Mailing List -
 Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] PRI
blocking on incoming calls





Ok.

First of all , be sure Redfone ethernet link and the
Asterisk ethernet link are both on the same switch segment.

Then try an pri intense debug on asterisk
console. I believe (not sure), this link is not at IP level but ethernet level
2

It can help to determine if packets get stucked into the
redphone or it is an issue related to Asterisk.



Hope it helps





Ing. Oscar Andrés Carriles

Presidente

InFoDaX Consultants

Nicolás Jorge 994 (B1706AVA) Haedo

Buenos Aires, Argentina

Tel: 54 11 4650 1775

Fax: 54 11 4650 4295

www.infodax.com.ar





-Mensaje original-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy
Enviado el: Martes, 18 de Abril de
2006 04:03 p.m.
Para: 'Asterisk Users Mailing List
 - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users] PRI
blocking on incoming calls



We have a crossover from
telco to the CSU and a crossover from the CSU to the RedFone and then a regular
Ethernet cable from the RedFone to the Asterisk.















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oscar Carriles
Sent: Tuesday, April 18, 2006 2:01
PM
To: 'Asterisk Users Mailing List -
 Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] PRI
blocking on incoming calls





I believe it is important
to determine if the issue arrives at TDMoE level (RedPhone uses it to avoid a
direct T1 link with the box ) 

Or at PRI level. I am not
clear what kink of link bridges your redPhone to Asterisk. Is it an ethernet
link or a T1 crossover?







Ing. Oscar Andrés
Carriles

Presidente

InFoDaX Consultants

Nicolás Jorge 994
(B1706AVA) Haedo

Buenos Aires,
Argentina

Tel: 54 11
4650 1775

Fax: 54 11 4650
4295

www.infodax.com.ar





-Mensaje original-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy
Enviado el: Martes, 18 de Abril de
2006 03:30 p.m.
Para:
asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] PRI
blocking on incoming calls



Ok here is our setup. We are using Asterisk 1.2.6 and
Zaptel 1.2.5. We are using RedFones FoneBridges. We also have a
Nortel Option 11C that we have hooked up to the Asterisk.



We have 3 T1s from MCI into one FoneBridge on
ports 1 to 3 using d4 and ami signaling. Then we have a local Qwest T1 on port
4 using esf and b8zs. All four are configured with em_w. We are, at this point,
only using Asterisk as an IVR with plans to move off the Nortel in the future
if we can make this work. We have a second FoneBridge with four PRIs
connected to our Nortel 11C using esf and b8zs and pri_net. The telco
T1s do not have D-Channels but the Nortel do. Calls come into the first
FoneBridge and into Asterisk. They are played a message about call recording
and then the call is transferred to the Nortel system to be processed by an
agent.



When we first fire this up all seems to work just
fine, calls come in, get the message and then transfer to the Nortel and on to
an agent. Everybody is happy.



The problem is after 5-20 minutes calls on the MCI
lines start getting busy signals. The Qwest line NEVER stops working. We would
place a few test calls on the MCI and get busy signals and then they start
going through again. A few minutes later they get busy signals again.



When we get the busy signals there is no response on
the Asterisk CLI with verbose at 10. Its as if the Asterisk is not ever
seeing the call.



What is annoying is that it works fine for a bit and
then starts hiccupping. 



Can anyone shed any light on where to look? Any help
would be desperately appreciated.



Please help.



_



Kevin Savoy

Business Unit Telecom Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark
of Novo 1, Inc










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[Asterisk-Users] PRI blocking on incoming calls

2006-04-18 Thread Kevin Savoy









Ok
here is our setup. We are using Asterisk 1.2.6 and Zaptel 1.2.5. We are using
RedFones FoneBridges. We also have a Nortel Option 11C that we
have hooked up to the Asterisk.



We
have 3 T1s from MCI into one FoneBridge on ports 1 to 3 using d4 and ami
signaling. Then we have a local Qwest T1 on port 4 using esf and b8zs. All four
are configured with em_w. We are, at this point, only using Asterisk as an IVR
with plans to move off the Nortel in the future if we can make this work. We
have a second FoneBridge with four PRIs connected to our Nortel 11C
using esf and b8zs and pri_net. The telco T1s do not have D-Channels but
the Nortel do. Calls come into the first FoneBridge and into Asterisk. They are
played a message about call recording and then the call is transferred to the
Nortel system to be processed by an agent.



When
we first fire this up all seems to work just fine, calls come in, get the
message and then transfer to the Nortel and on to an agent. Everybody is happy.



The
problem is after 5-20 minutes calls on the MCI lines start getting busy
signals. The Qwest line NEVER stops working. We would place a few test calls on
the MCI and get busy signals and then they start going through again. A few
minutes later they get busy signals again.



When
we get the busy signals there is no response on the Asterisk CLI with verbose
at 10. Its as if the Asterisk is not ever seeing the call.



What
is annoying is that it works fine for a bit and then starts hiccupping. 



Can
anyone shed any light on where to look? Any help would be desperately
appreciated.



Please
help.



_



Kevin Savoy

Business Unit Telecom
Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark of Novo 1, Inc








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RE: [Asterisk-Users] PRI blocking on incoming calls

2006-04-18 Thread Kevin Savoy









We have a crossover from telco to the CSU and a crossover
from the CSU to the RedFone and then a regular Ethernet cable from the RedFone
to the Asterisk.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oscar Carriles
Sent: Tuesday, April 18, 2006 2:01
PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] PRI
blocking on incoming calls





I believe it is important to determine if the issue arrives
at TDMoE level (RedPhone uses it to avoid a direct T1 link with the box ) 

Or at PRI level. I am not clear what kink of link bridges
your redPhone to Asterisk. Is it an ethernet link or a T1 crossover?







Ing. Oscar Andrés Carriles

Presidente

InFoDaX Consultants

Nicolás Jorge 994 (B1706AVA) Haedo

Buenos Aires, Argentina

Tel: 54 11 4650 1775

Fax: 54 11 4650 4295

www.infodax.com.ar





-Mensaje original-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy
Enviado el: Martes, 18 de Abril de
2006 03:30 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] PRI
blocking on incoming calls



Ok here is our setup. We are using Asterisk 1.2.6 and
Zaptel 1.2.5. We are using RedFones FoneBridges. We also have a
Nortel Option 11C that we have hooked up to the Asterisk.



We have 3 T1s from MCI into one FoneBridge on
ports 1 to 3 using d4 and ami signaling. Then we have a local Qwest T1 on port
4 using esf and b8zs. All four are configured with em_w. We are, at this point,
only using Asterisk as an IVR with plans to move off the Nortel in the future
if we can make this work. We have a second FoneBridge with four PRIs
connected to our Nortel 11C using esf and b8zs and pri_net. The telco
T1s do not have D-Channels but the Nortel do. Calls come into the first
FoneBridge and into Asterisk. They are played a message about call recording
and then the call is transferred to the Nortel system to be processed by an
agent.



When we first fire this up all seems to work just
fine, calls come in, get the message and then transfer to the Nortel and on to
an agent. Everybody is happy.



The problem is after 5-20 minutes calls on the MCI
lines start getting busy signals. The Qwest line NEVER stops working. We would
place a few test calls on the MCI and get busy signals and then they start
going through again. A few minutes later they get busy signals again.



When we get the busy signals there is no response on
the Asterisk CLI with verbose at 10. Its as if the Asterisk is not ever
seeing the call.



What is annoying is that it works fine for a bit and
then starts hiccupping. 



Can anyone shed any light on where to look? Any help
would be desperately appreciated.



Please help.



_



Kevin Savoy

Business Unit Telecom Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com

Novo 1 is a service mark
of Novo 1, Inc










--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006
 



--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006
 

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