date:20030905

On Fri, 2003-09-05 at 00:05, John Todd wrote:
 As has been noted before on this list, the Wait() application does
 not listen for keystrokes from users.  Many of you, like me, have
 looping Background(), Wait(), and Goto() application priority chains
 that prompt users to enter some data, and then repeat the
 instructions if no keys are pressed.  The problem of course is if the
 user doesn't start pressing keys during the Background() call and
 delays until the Wait() application is called, those keys are lost.
 I had solved this some time back by creating a few random length
 files of silence, that would replace Wait() routines in some
 circumstances.  I have finally created a formal measured group of
 files, each with 1-10 seconds of silence, and put them in my sounds
 directory for public consumption.  Not a big deal for most of you to
 create these files yourselves, but perhaps a minor pain that
 hopefully I've removed for some people who don't have sound tools
 handy.
 http://www.loligo.com/asterisk/sounds/silence/
And you missed the right way to deal with this. You may have to break
your extensions into more contexts, but you let the timeout function do
it's work.
[learning_the_way]
exten = s,1,DigitTimeout,5
exten = s,2,ResponseTimeout,10
exten = s,3,background(instructions)
exten = s,4,background(more_instructions)
exten = t,1,Goto(s|3)

This will let the call progress through the backgound apps, and if it
falls out of these rules, then it waits 10 seconds and falls into the t
extension where you can do whatever you like even going back and
repeating the instructions.
--
Steven Critchfield [EMAIL PROTECTED]
No, I didn't quite miss that method, but your example is useful in 
certain circumstances.

There are instances where you have pauses in between voice prompts 
that are not necessarily looped; I did not include in my message all 
possible iterations of why these silent gaps might be required, but 
since that is non-obvious, I include an instance below.  It could be 
done with multiple contexts or meta-extensions, but I feel that is 
inelegant and confusing when it need not be.  There are multiple ways 
to do things with Asterisk; not all of them fall into definitions of 
the right way for all administrators.

[nonpedanticexample]
exten = s,1,DigitTimeout(5)
exten = s,2,ResponseTimeout(20)
exten = s,3,Background(type-your-selection)
exten = s,4,Background(silence/3)
exten = s,5,Background(type-your-selection)
exten = s,6,Background(silence/3)
exten = s,7,Background(if-you-need-help-press-pound-for-instructions)
exten = t,1,Goto(s,3)

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Re: [Asterisk-Users] The sounds of silence: silent soundfiles available

On Fri, 2003-09-05 at 01:40, John Todd wrote:

 [nonpedanticexample]
 exten = s,1,DigitTimeout(5)
 exten = s,2,ResponseTimeout(20)
 exten = s,3,Background(type-your-selection)
 exten = s,4,Background(silence/3)
 exten = s,5,Background(type-your-selection)
 exten = s,6,Background(silence/3)
 exten = s,7,Background(if-you-need-help-press-pound-for-instructions)
 
 exten = t,1,Goto(s,3)


This is another way. Please think about limiting your loops like I have
below. It is possible to get a channel that didn't detect a hangup, and
would stay busy the way you have listed above. 

[anotherway]
exten = s,1,SetVar(Loop=0)
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(20)
exten = s,4,Background(type-your-selection)


exten = t,1,SetVar(Loop=[${Loop}+1])
exten = t,2,gotoif([${Loop}  6]?t|100)
exten = t,3,gotoif([${Loop} == 3]?t|200)
exten = t,4,goto(s|3)

exten = t,100,Hangup

exten = t,200,Background(if-you-need-help-press-pound-for-instructions)
exten = t,201,goto(s|3)



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[Asterisk-Users] Hardware IAX phone (please read and reply!)

2003-09-05 Thread Grzegorz Nosek

Hello all!

I've talked recently to the head of RD dept. of Telkom Telos 
(www.telos.com.pl) - a big Polish company specialised in making 
phones. I gave them the idea of creating a cheap (cost-effective) 
hardware IP phone. The phone we discussed would include hardware 
support for IAX (though probably SIP/sth. else would be required too 
if it were to hit the market.. what do you think?) and GSM 06.10. 
Although they have no previous experience in IP phones, they were 
quite interested and promised to have a deeper look into the issue. 

So now for the big part: everybody PLEASE give your suggestions about 
what the IP phone of your choice should look/work/... like. The main 
reason we started the talks was the cost of currently available 
phones (even $70+sh is a truckload of money for a phone here in 
Poland) but any and all suggestions are welcome.

I'd also love to hear from the more hardware-oriented people - do you 
have any suggestions about used chips, controllers, codecs, whatever? 
As I said, although they've been making phones for years, they 
haven't built an IP phone before so they have to research the 
possible elements used. Why not make it easier for them? :)

With Telos being a specialised factory, there's the benefit that e.g. 
good-looking cases are no problem at all, and if low price wasn't the 
goal, touchscreens and all would be an option too - maybe 
some deluxe edition?

An alternative design that came up was a bigger (say, 12/24 ports) 
gateway with some embedded Linux running on an industrial PC (as 
beefy as circumstances require - any comments?) with plain RJ11 
sockets on one side and Ethernet on the other. What do you think 
about this?

Hope to hear from you (a lot! :)

 Grzegorz Nosek
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[Asterisk-Users] Realm..

Is there an easy way to change the realm used for authentication from asterisk to 
anything else e.g. mydomain.com ??

Thanks
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Re: [Asterisk-Users] Hardware IAX phone (please read and reply!)

If there was a native IAX phone with GSM support and was around $70, I'd
buy a few, and I know several people in my social groups would get them.
I could even make a business case to get them for the office. 

On Fri, 2003-09-05 at 02:56, Grzegorz Nosek wrote:
 Hello all!
 
 I've talked recently to the head of RD dept. of Telkom Telos 
 (www.telos.com.pl) - a big Polish company specialised in making 
 phones. I gave them the idea of creating a cheap (cost-effective) 
 hardware IP phone. The phone we discussed would include hardware 
 support for IAX (though probably SIP/sth. else would be required too 
 if it were to hit the market.. what do you think?) and GSM 06.10. 
 Although they have no previous experience in IP phones, they were 
 quite interested and promised to have a deeper look into the issue. 
 
 So now for the big part: everybody PLEASE give your suggestions about 
 what the IP phone of your choice should look/work/... like. The main 
 reason we started the talks was the cost of currently available 
 phones (even $70+sh is a truckload of money for a phone here in 
 Poland) but any and all suggestions are welcome.
 
 I'd also love to hear from the more hardware-oriented people - do you 
 have any suggestions about used chips, controllers, codecs, whatever? 
 As I said, although they've been making phones for years, they 
 haven't built an IP phone before so they have to research the 
 possible elements used. Why not make it easier for them? :)
 
 With Telos being a specialised factory, there's the benefit that e.g. 
 good-looking cases are no problem at all, and if low price wasn't the 
 goal, touchscreens and all would be an option too - maybe 
 some deluxe edition?
 
 An alternative design that came up was a bigger (say, 12/24 ports) 
 gateway with some embedded Linux running on an industrial PC (as 
 beefy as circumstances require - any comments?) with plain RJ11 
 sockets on one side and Ethernet on the other. What do you think 
 about this?
 
 Hope to hear from you (a lot! :)
 
  Grzegorz Nosek
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Re: [Asterisk-Users] Hardware IAX phone (please read and reply!)

Sounds like a good idea..

My suggestion on looks and features would be to look at somthing like the Snom200, 
features like the ability to connect a standard pc type head set to the phone a great 
cost cutting features..

As for codecs I would look and G.711, G.729, GSM, iLBC and Speex.. That way you have 
covered both high and low bandwidth, closed source and open source codecs..

Look ate STUN support and possibly uPNP..

As for the rest just stick to the standards for SIP ( and IAX if you decied to 
impliment it)..

If the price per phone can be brought down closer to the cost of a standard analog 
phone then there will be no reason why VoIP will not take over the telecomunications 
of the world..

So I would say in the fisrt instance get a low cost, hight quality, reliable product 
created first and if there is a need then look at the Delux products.. Unfortunately 
the VoIP world is full of Delux but not enough budget products..

It seams that most only think of the global enterprise and not about the worlds SME's..

Later..

 Hello all!
 
 I've talked recently to the head of RD dept. of Telkom Telos 
 (www.telos.com.pl) - a big Polish company specialised in making 
 phones. I gave them the idea of creating a cheap (cost-effective) 
 hardware IP phone. The phone we discussed would include hardware 
 support for IAX (though probably SIP/sth. else would be required too 
 if it were to hit the market.. what do you think?) and GSM 06.10. 
 Although they have no previous experience in IP phones, they were 
 quite interested and promised to have a deeper look into the issue. 
 
 So now for the big part: everybody PLEASE give your suggestions about 
 what the IP phone of your choice should look/work/... like. The main 
 reason we started the talks was the cost of currently available 
 phones (even $70+sh is a truckload of money for a phone here in 
 Poland) but any and all suggestions are welcome.
 
 I'd also love to hear from the more hardware-oriented people - do you 
 have any suggestions about used chips, controllers, codecs, whatever? 
 As I said, although they've been making phones for years, they 
 haven't built an IP phone before so they have to research the 
 possible elements used. Why not make it easier for them? :)
 
 With Telos being a specialised factory, there's the benefit that e.g. 
 good-looking cases are no problem at all, and if low price wasn't the 
 goal, touchscreens and all would be an option too - maybe 
 some deluxe edition?
 
 An alternative design that came up was a bigger (say, 12/24 ports) 
 gateway with some embedded Linux running on an industrial PC (as 
 beefy as circumstances require - any comments?) with plain RJ11 
 sockets on one side and Ethernet on the other. What do you think 
 about this?
 
 Hope to hear from you (a lot! :)
 
  Grzegorz Nosek
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[Asterisk-Users] Transfer (again!)

2003-09-05 Thread Daniel ANDRE

Hello,

I am building an asterisk PBX with some stuff to make a usable VOIP / 
PSTN Gateway. I use the following devices:
SIP Phones from GrandStream for VOIP side
OpenLine4 from voicetronix for PSTN Side

I am building things step by step with some priorities.

I have now a working system able to place and receive calls from/to pstn.

Before attempting to bring other functions (like voice messaging) up i 
want to have a proper call transfert functionnality.
I can't have either blind transfert or consultative transfert working 
properly.  I am VERY interested in consultative transfert but I don't 
see where and how 'transfer', 'flash' or 'hold' keys and handle in 
asterisk code.

What I would like to do is:
A and B are taking each other
A press flash key: B listens music (thet works) and A can call C
A and C can talk each other but there is no mean for A to transfert B to 
C. Where should I patch the code to be able to do that?
Here A can talk either with B or C by pressing on 'Flash' Key but can't 
hang up any call.

IF C is Unavalaible, I haven't seen how to get B back

I welcome any idea about transfert application as it is a main issue for me:
AGI application,
Use of Transfert built in
Proper use of extension.conf file,
Patch to the source code of asterisk (I am able to do such a patch but I 
don't know where to look... chan_sip? apps directory, other?)

Best ragards,

Daniel

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IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
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Re: [Asterisk-Users] Hardware IAX phone (please read and reply!)

2003-09-05 Thread Marcel Prisi

I must say that I would be EXTREMELY interested in distributing such 
phones here in Switzerland ... We see a lot of demand here ... I am even 
willing to beta-test if needed.

For hardware/software infos, have a look at :

http://www.tuxscreen.net/

This is a completely open-source and open-hardware hardware phone based 
on Linux on an ARM embedded platform ... they already had lots of 
experience ... but might need some different software ...

Steven Critchfield wrote:

If there was a native IAX phone with GSM support and was around $70, I'd
buy a few, and I know several people in my social groups would get them.
I could even make a business case to get them for the office. 

On Fri, 2003-09-05 at 02:56, Grzegorz Nosek wrote:

Hello all!

I've talked recently to the head of RD dept. of Telkom Telos 
(www.telos.com.pl) - a big Polish company specialised in making 
phones. I gave them the idea of creating a cheap (cost-effective) 
hardware IP phone. The phone we discussed would include hardware 
support for IAX (though probably SIP/sth. else would be required too 
if it were to hit the market.. what do you think?) and GSM 06.10. 
Although they have no previous experience in IP phones, they were 
quite interested and promised to have a deeper look into the issue. 

So now for the big part: everybody PLEASE give your suggestions about 
what the IP phone of your choice should look/work/... like. The main 
reason we started the talks was the cost of currently available 
phones (even $70+sh is a truckload of money for a phone here in 
Poland) but any and all suggestions are welcome.

I'd also love to hear from the more hardware-oriented people - do you 
have any suggestions about used chips, controllers, codecs, whatever? 
As I said, although they've been making phones for years, they 
haven't built an IP phone before so they have to research the 
possible elements used. Why not make it easier for them? :)

With Telos being a specialised factory, there's the benefit that e.g. 
good-looking cases are no problem at all, and if low price wasn't the 
goal, touchscreens and all would be an option too - maybe 
some deluxe edition?

An alternative design that came up was a bigger (say, 12/24 ports) 
gateway with some embedded Linux running on an industrial PC (as 
beefy as circumstances require - any comments?) with plain RJ11 
sockets on one side and Ethernet on the other. What do you think 
about this?

Hope to hear from you (a lot! :)

Grzegorz Nosek
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Re: [Asterisk-Users] Transfer (again!)

These are probably more issues for grandstream.. Maybe mail [EMAIL PROTECTED] with the issues about dropping both calls when the phone is hung up..

Later

Hello,

I am building an asterisk PBX with some stuff to make a usable VOIP /
PSTN Gateway. I use the following devices:
SIP Phones from GrandStream for VOIP side
OpenLine4 from voicetronix for PSTN Side

I am building things step by step with some priorities.

I have now a working system able to place and receive calls from/to pstn.

Before attempting to bring other functions (like voice messaging) up i
want to have a proper call transfert functionnality.
I can't have either blind transfert or consultative transfert working
properly. I am VERY interested in consultative transfert but I don't
see where and how 'transfer', 'flash' or 'hold' keys and handle in
asterisk code.

What I would like to do is:
A and B are taking each other
A press flash key: B listens music (thet works) and A can call C
A and C can talk each other but there is no mean for A to transfert B to
C. Where should I patch the code to be able to do that?
Here A can talk either with B or C by pressing on 'Flash' Key but can't
hang up any call.

IF C is Unavalaible, I haven't seen how to get B back

I welcome any idea about transfert application as it is a main issue for me:
AGI application,
Use of Transfert built in
Proper use of extension.conf file,
Patch to the source code of asterisk (I am able to do such a patch but I
don't know where to look... chan_sip? apps directory, other?)

Best ragards,

Daniel

--
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IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
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Re: [Asterisk-Users] Transfer (again!)

2003-09-05 Thread William Zhang

GS phone does blind transfer only. Afer pressing transfer, you will
hear dialtone and then dial the number, after dial the whole number,
either wait more than 5 seconds or press redial/send button, then
hangup, it should work.

--- WipeOut . [EMAIL PROTECTED] wrote:
 These are probably more issues for grandstream.. Maybe mail
 [EMAIL PROTECTED] with the issues about dropping both calls
 when the phone is hung up..
 
 Later
 
  Hello,
  
  I am building an asterisk PBX with some stuff to make a usable VOIP
 / 
  PSTN Gateway. I use the following devices:
  SIP Phones from GrandStream for VOIP side
  OpenLine4 from voicetronix for PSTN Side
  
  I am building things step by step with some priorities.
  
  I have now a working system able to place and receive calls from/to
 pstn.
  
  Before attempting to bring other functions (like voice messaging)
 up i 
  want to have a proper call transfert functionnality.
  I can't have either blind transfert or consultative transfert
 working 
  properly.  I am VERY interested in consultative transfert but I
 don't 
  see where and how 'transfer', 'flash' or 'hold' keys and handle in 
  asterisk code.
  
  What I would like to do is:
  A and B are taking each other
  A press flash key: B listens music (thet works) and A can call C
  A and C can talk each other but there is no mean for A to transfert
 B to 
  C. Where should I patch the code to be able to do that?
  Here A can talk either with B or C by pressing on 'Flash' Key but
 can't 
  hang up any call.
  
  
  IF C is Unavalaible, I haven't seen how to get B back
  
  I welcome any idea about transfert application as it is a main
 issue for me:
  AGI application,
  Use of Transfert built in
  Proper use of extension.conf file,
  Patch to the source code of asterisk (I am able to do such a patch
 but I 
  don't know where to look... chan_sip? apps directory, other?)
  
  Best ragards,
  
  Daniel
  
  -- 
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  IRIS Technologies - http://www.iris-tech.com
  Serveur kwartz - http://www.kwartz.com
  
  
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=

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Re: [Asterisk-Users] Transfer (again!)

2003-09-05 Thread Daniel ANDRE





This works only if transfering to a phone wich is onhook. If it is off
hook (busy), it doesn't work

Is there any possibiliy to simulate transfert with dial plan?

Regards,
Daniel

William Zhang a crit:

  GS phone does blind transfer only. Afer pressing transfer, you will
hear dialtone and then dial the number, after dial the whole number,
either wait more than 5 seconds or press "redial/send" button, then
hangup, it should work.

--- "WipeOut ." [EMAIL PROTECTED] wrote:
  
  
These are probably more issues for grandstream.. Maybe mail
[EMAIL PROTECTED] with the issues about dropping both calls
when the phone is hung up..

Later



  Hello,

I am building an asterisk PBX with some stuff to make a usable VOIP
  

/ 


  PSTN Gateway. I use the following devices:
SIP Phones from GrandStream for VOIP side
OpenLine4 from voicetronix for PSTN Side

I am building things step by step with some priorities.

I have now a working system able to place and receive calls from/to
  

pstn.


  Before attempting to bring other functions (like voice messaging)
  

up i 


  want to have a proper call transfert functionnality.
I can't have either blind transfert or consultative transfert
  

working 


  properly.  I am VERY interested in consultative transfert but I
  

don't 


  see where and how 'transfer', 'flash' or 'hold' keys and handle in 
asterisk code.

What I would like to do is:
A and B are taking each other
A press flash key: B listens music (thet works) and A can call C
A and C can talk each other but there is no mean for A to transfert
  

B to 


  C. Where should I patch the code to be able to do that?
Here A can talk either with B or C by pressing on 'Flash' Key but
  

can't 


  hang up any call.


IF C is Unavalaible, I haven't seen how to get B back

I welcome any idea about transfert application as it is a main
  

issue for me:


  AGI application,
Use of Transfert built in
Proper use of extension.conf file,
Patch to the source code of asterisk (I am able to do such a patch
  

but I 


  don't know where to look... chan_sip? apps directory, other?)

Best ragards,

Daniel

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IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com


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=

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Re: [Asterisk-Users] Transfer (again!)

The problem I thought he was refering to was that if phoneA is in a call with phoneB, 
then phoneA uses flash to put phoneB on hold and call phoneC then..

Problem 1
If phoneC hangs up then both the calls from phoneA to phoneC and phoneA to phoneB are 
disconnected.

Problem 2
PhoneA has no way of disconecting phoneC and returning to phoneB. (flash button can be 
used to return to the call with phoneB but if phoneC doesn't hang up the call stays 
conencted)

later..


 GS phone does blind transfer only. Afer pressing transfer, you will
 hear dialtone and then dial the number, after dial the whole number,
 either wait more than 5 seconds or press redial/send button, then
 hangup, it should work.
 
 --- WipeOut . [EMAIL PROTECTED] wrote:
  These are probably more issues for grandstream.. Maybe mail
  [EMAIL PROTECTED] with the issues about dropping both calls
  when the phone is hung up..
  
  Later
  
   Hello,
   
   I am building an asterisk PBX with some stuff to make a usable VOIP
  / 
   PSTN Gateway. I use the following devices:
   SIP Phones from GrandStream for VOIP side
   OpenLine4 from voicetronix for PSTN Side
   
   I am building things step by step with some priorities.
   
   I have now a working system able to place and receive calls from/to
  pstn.
   
   Before attempting to bring other functions (like voice messaging)
  up i 
   want to have a proper call transfert functionnality.
   I can't have either blind transfert or consultative transfert
  working 
   properly.  I am VERY interested in consultative transfert but I
  don't 
   see where and how 'transfer', 'flash' or 'hold' keys and handle in 
   asterisk code.
   
   What I would like to do is:
   A and B are taking each other
   A press flash key: B listens music (thet works) and A can call C
   A and C can talk each other but there is no mean for A to transfert
  B to 
   C. Where should I patch the code to be able to do that?
   Here A can talk either with B or C by pressing on 'Flash' Key but
  can't 
   hang up any call.
   
   
   IF C is Unavalaible, I haven't seen how to get B back
   
   I welcome any idea about transfert application as it is a main
  issue for me:
   AGI application,
   Use of Transfert built in
   Proper use of extension.conf file,
   Patch to the source code of asterisk (I am able to do such a patch
  but I 
   don't know where to look... chan_sip? apps directory, other?)
   
   Best ragards,
   
   Daniel
   
   -- 
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   IRIS Technologies - http://www.iris-tech.com
   Serveur kwartz - http://www.kwartz.com
   
   
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Re: [Asterisk-Users] Transfer (again!)

2003-09-05 Thread Daniel ANDRE





It is exactly that and noway for PhoneA to connect PoneB and PhoneC
each other.

Daniel

WipeOut . a crit:

  The problem I thought he was refering to was that if phoneA is in a call with phoneB, then phoneA uses "flash" to put phoneB on hold and call phoneC then..

Problem 1
If phoneC hangs up then both the calls from phoneA to phoneC and phoneA to phoneB are disconnected.

Problem 2
PhoneA has no way of disconecting phoneC and returning to phoneB. (flash button can be used to return to the call with phoneB but if phoneC doesn't hang up the call stays conencted)

later..


  
  
GS phone does blind transfer only. Afer pressing transfer, you will
hear dialtone and then dial the number, after dial the whole number,
either wait more than 5 seconds or press "redial/send" button, then
hangup, it should work.

--- "WipeOut ." [EMAIL PROTECTED] wrote:


  These are probably more issues for grandstream.. Maybe mail
[EMAIL PROTECTED] with the issues about dropping both calls
when the phone is hung up..

Later

  
  
Hello,

I am building an asterisk PBX with some stuff to make a usable VOIP

  
  / 
  
  
PSTN Gateway. I use the following devices:
SIP Phones from GrandStream for VOIP side
OpenLine4 from voicetronix for PSTN Side

I am building things step by step with some priorities.

I have now a working system able to place and receive calls from/to

  
  pstn.
  
  
Before attempting to bring other functions (like voice messaging)

  
  up i 
  
  
want to have a proper call transfert functionnality.
I can't have either blind transfert or consultative transfert

  
  working 
  
  
properly.  I am VERY interested in consultative transfert but I

  
  don't 
  
  
see where and how 'transfer', 'flash' or 'hold' keys and handle in 
asterisk code.

What I would like to do is:
A and B are taking each other
A press flash key: B listens music (thet works) and A can call C
A and C can talk each other but there is no mean for A to transfert

  
  B to 
  
  
C. Where should I patch the code to be able to do that?
Here A can talk either with B or C by pressing on 'Flash' Key but

  
  can't 
  
  
hang up any call.


IF C is Unavalaible, I haven't seen how to get B back

I welcome any idea about transfert application as it is a main

  
  issue for me:
  
  
AGI application,
Use of Transfert built in
Proper use of extension.conf file,
Patch to the source code of asterisk (I am able to do such a patch

  
  but I 
  
  
don't know where to look... chan_sip? apps directory, other?)

Best ragards,

Daniel

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IRIS Technologies - http://www.iris-tech.com
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Re: [Asterisk-Users] cisco ATA186 I2 vs I1

2003-09-05 Thread Florian Overkamp

At 00:56 5-9-2003 -0400, you wrote:
I saw your posting about the cisco ata186 I2 vs I1 and the simple vs 
complex impedance.
I ordered a cisco ata186 i2 for use in Canada by mistake, didn't know that 
I needed the I1
version.
Will the I2 version work in Canada with regular anlog phones, or will I 
need to change it.
Many modern analog phones can do both modes without too much problems, so 
chances are you won't notice any difference at all. However, some older 
equipment may not work.

Florian

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Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems ..

2003-09-05 Thread Thilo Salmon

On Thu, 2003-09-04 at 19:20, Dave Alan Caruana wrote:
 has anyone got G729 and SIP working together?
 some config examples would help :)

This configuration works for me:

sip.conf:

[grandstream] 
type=friend 
username=grandstream 
insecure=yes 
host=dynamic 
context=sip-out 
nat=yes 
canreinvite=no 
disallow=all 
allow=g729

extensions.conf:

[sip-out] 
exten = _.,1,Dial,Technology/Resource

Thilo


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RE: [Asterisk-Users] 7960 backup proxy registration

2003-09-05 Thread Rich Adamson

I'm no where near an expert (or even very knowledgable on some of this stuff),
but a fair number of machines (regardless of whether its a 7960 or whatever)
will not fail over to secondary/backup gateways unless the primary is totally
non-responsive. That usually means if the proxy responds with even an icmp
port unreachable, it is still responding and the phone won't fail over to the
backup. To validate, I'd suggest disconnecting the primary proxy to see if the
phone then registers with other servers.

Also, the v4.4 release notes for Open Caveats says... CSCea15061: Outbound
Proxy reREGISTER fails due to incorret logic. The Resolved Caveats tend to
suggest that some related problems were fixed in v4.4, so this must have been
an issue with previous releases.

 -Original Message-
 I'm sorry to ask this question, but I thought I'd rather ask it here before
 messing up with cisco.
 Is anybody running cisco 7960 in redundant configuration?
 I mean I want the phone to be registered with both primary and
 backup proxy (asterisks) so that service continues to work in case of
 primary
 proxy failure. I've set in SIPDefault.cnt:
 
 proxy1_address: 192.168.1.10
 proxy1_port: 5060
 proxy_backup: 192.168.1.12
 proxy_backup_port: 5060
 
 The problem is that 7960 registers all the configured lines with
 primary proxy, but the line 1 only with backup proxy. It's not about
 registration
 failure. The phone doesn't even try to register other 5 lines. As a result
 if the primary
 proxy fails incoming calls work for the line 1 only.
 Has anybody managed to register all the lines with backup proxy?
 I'm running software 4.4, the last version before digital signature was
 introduced. Should I upgrade? Or may be I'm missing something in
 configuration?


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[Asterisk-Users] Problems setting asterisk environment varibles

2003-09-05 Thread Carlos Fernández Puente

Title: Carlos Fernández Puente




Hi,
I have a
problemwhen i try to set an asterisk environment variable while asterisk
is running an AGI aplication.
I postthe
few code lines (in C).
printf ("SET
VARIABLE agisel = %s\n\n",agiselected);fprintf(stderr,"SET VARIABLE agisel =
%s\n\n",agiselected);
fprintf(stdout,"GET
VARIABLE agisel\n\n");

when i try to get
the response from asterisk it send me an error message like this.
510 Invalid or
unknown command

Can you help me
with this trouble?

thanks

Carlos Fernández
Puente [EMAIL PROTECTED]

Ingeniero de
proyectosAlisysSoftware




Alisys Software,
S.L.
Edificio Lexington - C/ Orense, 85 28020 MADRID Tfno.: 985175935 -
915678474 Fax: 915714244 web: http://www.alisys.net wap: http://www.alisys.net/wap/

[Asterisk-Users] call parking -- what was the key combination?

2003-09-05 Thread Dave Alan Caruana

hi great gurus of asterisk :)

could somebody remind me the key combination to send a call
into the parking queue ?

while you're at it, are there any other key combinations I should know??
eg. put a call on hold etc.

thanks
Dave



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RE: [Asterisk-Users] Call script after hangup

2003-09-05 Thread Frank N.

That makes perfect sense. It works perfectly.
Thanks to you and Matteo who suggested the same solution.


-Original Message-
From: Alastair Maw [mailto:[EMAIL PROTECTED]
Sent: 4 septembre, 2003 11:28
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call script after hangup

Frank N. wrote:
 I believe the porblem is that, since the incoming call is not closed
 before the outgoing call is created, the outgoing call does not work.
 I was hoping the delay would solve this problem... but obviously it
doesn't.
No - it still won't relinquish the call until the hangup handler has
completed. What you need to do is to have the AGI script return, such
that the call exits. Then five seconds later, copy the file.
You could do this by setting up a BASH script which executed the Perl in
the background. I.e.
#!/bin/sh
/path/to/script/foo.pl 
Make sense?
-- 
Alastair Maw [EMAIL PROTECTED]
MX Telecom - Systems Analyst
http://www.mxtelecom.com
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Re: [Asterisk-Users] call parking -- what was the key combination?

To park a call you simply transfer the call into extension 700 (this is the default 
and can be changed)..

To get the call back you just dial the parked location.. If you are using an IP phone 
this is a problem becasue it will not tell you the location of the parked call so you 
will not know where to collect it from..



 hi great gurus of asterisk :)
 
 could somebody remind me the key combination to send a call
 into the parking queue ?
 
 while you're at it, are there any other key combinations I should know??
 eg. put a call on hold etc.
 
 thanks
 Dave
 
 
 
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Re: [Asterisk-Users] Hardware IAX phone (please read and reply!)

On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote:
 I must say that I would be EXTREMELY interested in distributing such 
 phones here in Switzerland ... We see a lot of demand here ... I am even 
 willing to beta-test if needed.
 
 For hardware/software infos, have a look at :
 
 http://www.tuxscreen.net/
 
 This is a completely open-source and open-hardware hardware phone based 
 on Linux on an ARM embedded platform ... they already had lots of 
 experience ... but might need some different software ...


bzzzt. wrong. There is a lot known about the hardware but it is not
open. The software is only open after it was reloaded with debian. Also
while the site you list was cheap, if you dig round, the manufacturing
cost was over $300 each and target retail was over %600. Granted that
was over 3 years ago, it wouldn't have dropped in price too
significantly. The site you list was liquidating the last known
inventory of those units. 


 Steven Critchfield wrote:
 
  If there was a native IAX phone with GSM support and was around $70, I'd
  buy a few, and I know several people in my social groups would get them.
  I could even make a business case to get them for the office. 
  
  On Fri, 2003-09-05 at 02:56, Grzegorz Nosek wrote:
  
 Hello all!
 
 I've talked recently to the head of RD dept. of Telkom Telos 
 (www.telos.com.pl) - a big Polish company specialised in making 
 phones. I gave them the idea of creating a cheap (cost-effective) 
 hardware IP phone. The phone we discussed would include hardware 
 support for IAX (though probably SIP/sth. else would be required too 
 if it were to hit the market.. what do you think?) and GSM 06.10. 
 Although they have no previous experience in IP phones, they were 
 quite interested and promised to have a deeper look into the issue. 
 
 So now for the big part: everybody PLEASE give your suggestions about 
 what the IP phone of your choice should look/work/... like. The main 
 reason we started the talks was the cost of currently available 
 phones (even $70+sh is a truckload of money for a phone here in 
 Poland) but any and all suggestions are welcome.
 
 I'd also love to hear from the more hardware-oriented people - do you 
 have any suggestions about used chips, controllers, codecs, whatever? 
 As I said, although they've been making phones for years, they 
 haven't built an IP phone before so they have to research the 
 possible elements used. Why not make it easier for them? :)
 
 With Telos being a specialised factory, there's the benefit that e.g. 
 good-looking cases are no problem at all, and if low price wasn't the 
 goal, touchscreens and all would be an option too - maybe 
 some deluxe edition?
 
 An alternative design that came up was a bigger (say, 12/24 ports) 
 gateway with some embedded Linux running on an industrial PC (as 
 beefy as circumstances require - any comments?) with plain RJ11 
 sockets on one side and Ethernet on the other. What do you think 
 about this?
 
 Hope to hear from you (a lot! :)
 
  Grzegorz Nosek
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Re: [Asterisk-Users] Hardware IAX phone (please read and reply!)

2003-09-05 Thread Marcel Prisi

Steven Critchfield wrote:

On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote:

I must say that I would be EXTREMELY interested in distributing such 
phones here in Switzerland ... We see a lot of demand here ... I am even 
willing to beta-test if needed.

For hardware/software infos, have a look at :

http://www.tuxscreen.net/

This is a completely open-source and open-hardware hardware phone based 
on Linux on an ARM embedded platform ... they already had lots of 
experience ... but might need some different software ...


bzzzt. wrong. There is a lot known about the hardware but it is not
open. The software is only open after it was reloaded with debian. Also
while the site you list was cheap, if you dig round, the manufacturing
cost was over $300 each and target retail was over %600. Granted that
was over 3 years ago, it wouldn't have dropped in price too
significantly. The site you list was liquidating the last known
inventory of those units. 

So have a look there :

http://www.lart.tudelft.nl/

You will find there the hardware that evolved from what was in the 
Tuxscreen. It's license is open. It runs a 220Mhz StrongARM with more 
than 200 MIPS and has options for ethernet and sound i/o, all is 
linux-compatible ...

Maybe useful for prototyping ?

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Re: [Asterisk-Users] Hardware IAX phone (please read and reply!)

On Fri, 2003-09-05 at 08:58, Marcel Prisi wrote:
 Steven Critchfield wrote:
 
  On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote:
  
 I must say that I would be EXTREMELY interested in distributing such 
 phones here in Switzerland ... We see a lot of demand here ... I am even 
 willing to beta-test if needed.
 
 For hardware/software infos, have a look at :
 
 http://www.tuxscreen.net/
 
 This is a completely open-source and open-hardware hardware phone based 
 on Linux on an ARM embedded platform ... they already had lots of 
 experience ... but might need some different software ...
  
  
  
  bzzzt. wrong. There is a lot known about the hardware but it is not
  open. The software is only open after it was reloaded with debian. Also
  while the site you list was cheap, if you dig round, the manufacturing
  cost was over $300 each and target retail was over %600. Granted that
  was over 3 years ago, it wouldn't have dropped in price too
  significantly. The site you list was liquidating the last known
  inventory of those units. 
  
 
 So have a look there :
 
 http://www.lart.tudelft.nl/
 
 You will find there the hardware that evolved from what was in the 
 Tuxscreen. It's license is open. It runs a 220Mhz StrongARM with more 
 than 200 MIPS and has options for ethernet and sound i/o, all is 
 linux-compatible ...
 
 Maybe useful for prototyping ?

The kits would be over $300US and don't have a case or software loaded
on, nor a phone interface. Granted it is a decent starting point.
-- 
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[Asterisk-Users] oh323 call segmentation fault

2003-09-05 Thread Marian Danisek

hello,
i have problem with oh323 channel driver (tryied differnet versions).
when i try to make call between oh323 - sip, oh323-isdn, oh323-capi
asterisk crash with segmentation fault. Channel driver was compiled with
pwlib 1.5.0 and openh323 1.12.0 libs.
Does anybody know solution ?

WrapH323Connection::WrapH323Connection: WrapH323Connection created.
-- Executing Dial(H323:31119, SIP/92) in new stack
-- Called 92
-- SIP/92-e46b is ringing
-- SIP/92-e46b is ringing
-- SIP/92-e46b is ringing
-- SIP/92-e46b is ringing
-- SIP/92-e46b answered H323:31119
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
  0:58.180 H245:8128d60 RTP_UDP No mediaControlChannel
specified
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
Segmentation fault

regads 

Marian


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Tel: +421-46-5430 754 # Fax: +421-46-5439 144
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Re: [Asterisk-Users] Regular expression matching for : - examples needed

 Examples I'd like to see:

 1)
   ${FOO} contains 12345#
   ${HASH} contains #

something like this:

exten = 123,1,Gotoif($[${FOO} : 12345#]?2|102)


   If ${FOO} contains the contents of ${HASH} anywhere, go to 2. If not, goto 102

 exten= 123,1,GotoIf($[...???...]?2|102)


 1.1)
If the last digit of ${FOO} is ${HASH}, then goto 2.  If not, goto 102.


 exten = 123,1,GotoIf($[...???...]?2|102)

exten = 123,1,GotoIf($[${FOO:-1:1} = ${HASH}]?2|102)
assuming ${HASH} is one digit ...


Martin

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Re: [Asterisk-Users] The sounds of silence: silent soundfiles available

You could use ResponseTimeout together with Background instead of playing
silence files.

Martin

On Thu, 4 Sep 2003, John Todd wrote:


 As has been noted before on this list, the Wait() application does
 not listen for keystrokes from users.  Many of you, like me, have
 looping Background(), Wait(), and Goto() application priority chains
 that prompt users to enter some data, and then repeat the
 instructions if no keys are pressed.  The problem of course is if the
 user doesn't start pressing keys during the Background() call and
 delays until the Wait() application is called, those keys are lost.

 I had solved this some time back by creating a few random length
 files of silence, that would replace Wait() routines in some
 circumstances.  I have finally created a formal measured group of
 files, each with 1-10 seconds of silence, and put them in my sounds
 directory for public consumption.  Not a big deal for most of you to
 create these files yourselves, but perhaps a minor pain that
 hopefully I've removed for some people who don't have sound tools
 handy.

 http://www.loligo.com/asterisk/sounds/silence/

 JT
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Re: [Asterisk-Users] call parking -- what was the key combination?

2003-09-05 Thread Dave Alan Caruana

what i'm asking is what is the key sequence
you have to dial for the transfer ..

it was something like *7# if I remember
well, I know I had it working, but the client
lost the paper I wrote it on for him, and I can't
trace the email I got it from!

cheers
Dave

- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 05, 2003 3:11 PM
Subject: Re: [Asterisk-Users] call parking -- what was the key combination?


 To park a call you simply transfer the call into extension 700 (this is
the default and can be changed)..

 To get the call back you just dial the parked location.. If you are using
an IP phone this is a problem becasue it will not tell you the location of
the parked call so you will not know where to collect it from..



  hi great gurus of asterisk :)
 
  could somebody remind me the key combination to send a call
  into the parking queue ?
 
  while you're at it, are there any other key combinations I should know??
  eg. put a call on hold etc.
 
  thanks
  Dave
 
 
 
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Re: [Asterisk-Users] 7960 backup proxy registration

2003-09-05 Thread Michael Ulitskiy

Well, on the other hand Release Notes for software 4.2
(http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/relnote/phnrn42s.htm#58498)
says:
The SIP phone can register with a backup proxy to support Survivable
Remote Site Telephony (SRST). If the main proxy goes down, the backup
proxy has the registration information required to route calls successfully.

It does register the 1st line. It makes absolutely no sense to me to register
just the 1st line and abondone the others.

Michael

On Thursday 04 September 2003 11:38 pm, Shawn L. Djernes wrote:
From What I understand of this feature it is only to keep the phone working
not to provide full services. I think they intended it to be something like
a less powerful router or a box at a remote site. This way if the primary
server was took out by a virus or hardware failure your office staff could
still call for help.

Shawn

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael Ulitskiy
Sent: Thursday, September 04, 2003 18:10
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] 7960 backup proxy registration

Hi,

I'm sorry to ask this question, but I thought I'd rather ask it here before
messing up with cisco.
Is anybody running cisco 7960 in redundant configuration?
I mean I want the phone to be registered with both primary and
backup proxy (asterisks) so that service continues to work in case of
primary
proxy failure. I've set in SIPDefault.cnt:

proxy1_address: 192.168.1.10
proxy1_port: 5060
proxy_backup: 192.168.1.12
proxy_backup_port: 5060

The problem is that 7960 registers all the configured lines with
primary proxy, but the line 1 only with backup proxy. It's not about
registration
failure. The phone doesn't even try to register other 5 lines. As a result
if the primary
proxy fails incoming calls work for the line 1 only.
Has anybody managed to register all the lines with backup proxy?
I'm running software 4.4, the last version before digital signature was
introduced. Should I upgrade? Or may be I'm missing something in
configuration?
Thanks a lot.

Michael

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Re: [Asterisk-Users] call parking -- what was the key combination?

It's defined in /etc/asterisk/parking.conf

and set by deafult as 700

Martin

On Fri, 5 Sep 2003, Dave Alan Caruana wrote:

 what i'm asking is what is the key sequence
 you have to dial for the transfer ..

 it was something like *7# if I remember
 well, I know I had it working, but the client
 lost the paper I wrote it on for him, and I can't
 trace the email I got it from!

 cheers
 Dave

 - Original Message -
 From: WipeOut . [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Friday, September 05, 2003 3:11 PM
 Subject: Re: [Asterisk-Users] call parking -- what was the key combination?


  To park a call you simply transfer the call into extension 700 (this is
 the default and can be changed)..
 
  To get the call back you just dial the parked location.. If you are using
 an IP phone this is a problem becasue it will not tell you the location of
 the parked call so you will not know where to collect it from..
 
 
 
   hi great gurus of asterisk :)
  
   could somebody remind me the key combination to send a call
   into the parking queue ?
  
   while you're at it, are there any other key combinations I should know??
   eg. put a call on hold etc.
  
   thanks
   Dave
  
  
  
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Re: [Asterisk-Users] 7960 backup proxy registration

2003-09-05 Thread Michael Ulitskiy

On Friday 05 September 2003 08:21 am, Rich Adamson wrote:
 I'm no where near an expert (or even very knowledgable on some of this stuff),
 but a fair number of machines (regardless of whether its a 7960 or whatever)
 will not fail over to secondary/backup gateways unless the primary is totally
 non-responsive. That usually means if the proxy responds with even an icmp
 port unreachable, it is still responding and the phone won't fail over to the
 backup. To validate, I'd suggest disconnecting the primary proxy to see if the
 phone then registers with other servers.

No, it's not the case. The phone seems to work properly. It recognizes primary 
proxy failure and send INVITEs to the backup proxy. It does it for all lines.
It just doesn't register lines 2-6 with backup proxy, so inbound calls for the
numbers configured on those lines fail.
I was hoping that somebody more experienced than myself could confirm or 
refute it. May be somebody tried it with the latest software releases?
 
 Also, the v4.4 release notes for Open Caveats says... CSCea15061: Outbound
 Proxy reREGISTER fails due to incorret logic. The Resolved Caveats tend to
 suggest that some related problems were fixed in v4.4, so this must have been
 an issue with previous releases.

Well, I guess outbound proxy is a different story.
Thanks anyway.

Michael 

  -Original Message-
  I'm sorry to ask this question, but I thought I'd rather ask it here before
  messing up with cisco.
  Is anybody running cisco 7960 in redundant configuration?
  I mean I want the phone to be registered with both primary and
  backup proxy (asterisks) so that service continues to work in case of
  primary
  proxy failure. I've set in SIPDefault.cnt:
  
  proxy1_address: 192.168.1.10
  proxy1_port: 5060
  proxy_backup: 192.168.1.12
  proxy_backup_port: 5060
  
  The problem is that 7960 registers all the configured lines with
  primary proxy, but the line 1 only with backup proxy. It's not about
  registration
  failure. The phone doesn't even try to register other 5 lines. As a result
  if the primary
  proxy fails incoming calls work for the line 1 only.
  Has anybody managed to register all the lines with backup proxy?
  I'm running software 4.4, the last version before digital signature was
  introduced. Should I upgrade? Or may be I'm missing something in
  configuration?
 
 
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Re: [Asterisk-Users] oh323 call segmentation fault

2003-09-05 Thread Michael Ulitskiy

If you are using ulaw codec, try change it to alaw. 
oh323 currently has some problems with ulaw codec. 

Michael

On Friday 05 September 2003 10:22 am, Marian Danisek wrote:
 hello,
 i have problem with oh323 channel driver (tryied differnet versions).
 when i try to make call between oh323 - sip, oh323-isdn, oh323-capi
 asterisk crash with segmentation fault. Channel driver was compiled with
 pwlib 1.5.0 and openh323 1.12.0 libs.
 Does anybody know solution ?
 
 WrapH323Connection::WrapH323Connection: WrapH323Connection created.
 -- Executing Dial(H323:31119, SIP/92) in new stack
 -- Called 92
 -- SIP/92-e46b is ringing
 -- SIP/92-e46b is ringing
 -- SIP/92-e46b is ringing
 -- SIP/92-e46b is ringing
 -- SIP/92-e46b answered H323:31119
 PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
 PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
 PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
   0:58.180 H245:8128d60 RTP_UDP No mediaControlChannel
 specified
 PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
 PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
 PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
 Segmentation fault
 
 regads 
 
 Marian
 
 
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RE: [Asterisk-Users] call parking -- what was the key combination?

2003-09-05 Thread Andy Hester

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Dave Alan
 Caruana
 Sent: Friday, September 05, 2003 9:37 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] call parking -- what was the key
 combination?

 what i'm asking is what is the key sequence
 you have to dial for the transfer ..

 it was something like *7# if I remember
 well, I know I had it working, but the client
 lost the paper I wrote it on for him, and I can't
 trace the email I got it from!

 cheers
 Dave

I think its just # and then dial the number for parking ie #700

Andy

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[Asterisk-Users] IAX sound probs

2003-09-05 Thread Thomas Haeger

Hi all together,

i have following configuration:

ISDN Phone --- ASTERISK1/PRI --- ASTERISK1/IAX --- INTERNET ---INTERNET
ROUTER (Port 5036 nat) --- ASTERISK2/FXO/ANALOG DEV

The call flows fine, but no sound will be transfered.

On ASTERISK1 a message like stopped sounds occurs.


What' s wrong? Is there another port wich i have to nat ?


Regards, thanks for help,

Thomas.



***
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[Asterisk-Users] Manager / Windows Apps / Line Appearances

2003-09-05 Thread Steve Creel

It just dawned on me as I was playing with the manager interface - it
can't be very difficult at all to write an Win32 app that serves as a
lamp field.  Between 'Newchannel', 'Newstate', and 'Hangup' events, all
of the information is there.

I've heard several requests for line appearances, but mgcp and sccp
channels don't currently include support.  I know that in all the
instances I'd like to have call appearances, a windows application would
be an equally valid solution.

My problem is that I know nothing about writing little Win32 apps like
that.  While I can give it a shot (and I will), I'm sure there is someone
far more qualified who could probably write it much better and far more
quickly.

Just my $0.02

Steve


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Re: [Asterisk-Users] call parking -- what was the key combination?

2003-09-05 Thread James Sizemore

If you put  Tt in your dial statement you can type # some number to 
transfer to.
Of if you can send flash hooks that will work as well.

Dave Alan Caruana wrote:

what i'm asking is what is the key sequence
you have to dial for the transfer ..
it was something like *7# if I remember
well, I know I had it working, but the client
lost the paper I wrote it on for him, and I can't
trace the email I got it from!
cheers
Dave
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 05, 2003 3:11 PM
Subject: Re: [Asterisk-Users] call parking -- what was the key combination?
 

To park a call you simply transfer the call into extension 700 (this is
   

the default and can be changed)..
 

To get the call back you just dial the parked location.. If you are using
   

an IP phone this is a problem becasue it will not tell you the location of
the parked call so you will not know where to collect it from..
 

   

hi great gurus of asterisk :)

could somebody remind me the key combination to send a call
into the parking queue ?
while you're at it, are there any other key combinations I should know??
eg. put a call on hold etc.
thanks
Dave


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[Asterisk-Users] Windows 2000 call viewer!

2003-09-05 Thread Ariel Batista




I am new to this forum. As well as a new user 
of Asterisk. My vendor installed the system and we are still trying to get 
all the bugs out of it! I have a few questions about configuration and a 
program to view who is on what extensions.

I am looking for a program that will work on my 
Receptionist work station. She is running Windows 2000 pro. We have 
not plans on upgrading to XP pro so it's not an option at this time! We 
need to get asmall program that will lether 
viewwhoextensions are in use!

2nd problem is we have lost the caller ID 
function. We haveset the Zapata.conf to:

usecallerid = yes
hidecallerid = no

. But we are not getting any thing in. 
Internally we are getting extensions ID'slike if extension 114 calls 152 
they see our name and extension on there set!We are also not 
able to set the caller ID for outbound calls on our PRI lines! Here is a 
sample on how we use this settings in extensions.conf


exten= 
_91NXXNXX,1,SetCallerID(305XXX)
exten = _91NXXNXX,2,Dial (${LDTRUNK} / $ 
{EXTEN:1})
exten = _91NXXNXX,3,Congestion

Thank you in advance for any help with the 2 above 
problems.

Ariel BatistaAvionica, Inc.14380 SW 139 
Ct.Miami, FL 33186Ph: 305-256-0429 x114Fx: 305-574-0212web: http://www.avionica.comemail: [EMAIL PROTECTED]

[Asterisk-Users] Polycom IP Phones

2003-09-05 Thread Kevin Thompson

Does any one have any experience setting up asterisk with polycom IP 
phones? All i have been able to figure out about them is that they connect 
to an FTP site on boot. I tried going to the site to see what files are 
there but it seems they deny directory browsing.

Any one have any clues as to how i could configure my polycom IP phones?

thanks

Kevin

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[Asterisk-Users] CDR billable seconds

2003-09-05 Thread Steven Poelmans

Hello all,

I have a newbie question about the CDR.
Does billable seconds equal end time minus the time that a human actually picks up 
the phone?

thanks,
Steven

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Re: [Asterisk-Users] CDR billable seconds

2003-09-05 Thread Brancaleoni Matteo

yes, is to say that's equal to end time minus the ring time
before the remote party picked up.

matteo.

Il ven, 2003-09-05 alle 20:32, Steven Poelmans ha scritto:
 Hello all,
 
 I have a newbie question about the CDR.
 Does billable seconds equal end time minus the time that a human actually picks 
 up the phone?
 
 thanks,
 Steven
 
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[Asterisk-Users] Re: Asterisk Jitters

2003-09-05 Thread Zak






Hi Steven.

I have done as you suggested and I'm still getting the same problem.
/proc/interrupts lists the following:

0: 45489 XT-PIC timer
 1: 235 XT-PIC keyboard
 2: 0 XT-PIC cascade
 5: 335816 XT-PIC wcfxo, Intel ICH2
 8: 1 XT-PIC rtc
 9: 0 XT-PIC usb-uhci
10: 829 XT-PIC eth0
11: 0 XT-PIC usb-uhci
12: 194 XT-PIC PS/2 Mouse
14: 4402 XT-PIC ide0
15: 2 XT-PIC ide1
NMI: 0
ERR: 0

I am also getting the following message when asterisk starts.. but I'm
not sure if it means anything?

WARNING[16384]: File chan_oss.c, Line 974 (load_module): XXX I don't
work right
with non-full
duplex sound cards XXX
 == Registered channel type 'Console' (OSS Console Channel Driver)
 == Parsing '/etc/asterisk/oss.conf': Found
WARNING[114696]: File chan_oss.c, Line 232 (sound_thread): Read error
on sound
device: Resource temporarily unavailabl

thank,

Zak

Bing,Bing,Bing, we have the problem. nvidia and wcfxo cards on the same
interupt.

I'd say try removing a 2 WCFXO cards from the system and see if the
interupts free up, and your jitter stops.


  12: 524504  XT-PIC  PS/2 Mouse
  14: 165140  XT-PIC  ide0
  15: 281208  XT-PIC  ide1
 NMI:  0
 ERR:  0

Re: [Asterisk-Users] Re: Asterisk Jitters

2003-09-05 Thread Eric Wieling

As you can see wcfxo is still sharing an IRQ.  It won't work well if it
shares an IRQ.

On Fri, 2003-09-05 at 19:39, Zak wrote:
 Hi Steven.
 
 I have done as you suggested and I'm still getting the same problem.
 /proc/interrupts lists the following:
 
 0:  45489  XT-PIC  timer
   1:235  XT-PIC  keyboard
   2:  0  XT-PIC  cascade
   5: 335816  XT-PIC  wcfxo, Intel ICH2
   8:  1  XT-PIC  rtc
   9:  0  XT-PIC  usb-uhci
  10:829  XT-PIC  eth0
  11:  0  XT-PIC  usb-uhci
  12:194  XT-PIC  PS/2 Mouse
  14:   4402  XT-PIC  ide0
  15:  2  XT-PIC  ide1
 NMI:  0
 ERR:  0
 
 I am also getting the following message when asterisk starts.. but I'm
 not sure if it means anything?
 
 WARNING[16384]: File chan_oss.c, Line 974 (load_module): XXX I don't
 work right
 with non-full
 duplex sound cards XXX
   == Registered channel type 'Console' (OSS Console Channel Driver)
   == Parsing '/etc/asterisk/oss.conf': Found
 WARNING[114696]: File chan_oss.c, Line 232 (sound_thread): Read error
 on sound
 device: Resource temporarily unavailabl
 
 thank,
 
 Zak
 Bing,Bing,Bing, we have the problem. nvidia and wcfxo cards on the same
 interupt.
 
 I'd say try removing a 2 WCFXO cards from the system and see if the
 interupts free up, and your jitter stops.
 
   12: 524504  XT-PIC  PS/2 Mouse
   14: 165140  XT-PIC  ide0
   15: 281208  XT-PIC  ide1
  NMI:  0
  ERR:  0
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Re: [Asterisk-Users] X100P in Spain Busy Detect

2003-09-05 Thread Norberto Garcia Prieto

Martin Pycko wrote:

What's the Spain busy tone ? x ms tone, y ms of silence etc ...

 

   If I remember correctly, 0.2 ms on 0.2 ms off repeated. All tones 
are 425 Hz, -10dBm
It may also add 0.4ms off after every 3 on/off cycles

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Re: [Asterisk-Users] X100P in Spain Busy Detect

If you have 0.4 ms silence every 3 cycles then try to uncommnet
BUSYDETECT_TONEONLY in asterisk/Makefile and recompile.

regards
Martin

On Fri, 5 Sep 2003, Norberto Garcia Prieto wrote:

 Martin Pycko wrote:

 What's the Spain busy tone ? x ms tone, y ms of silence etc ...
 
 
 
 If I remember correctly, 0.2 ms on 0.2 ms off repeated. All tones
 are 425 Hz, -10dBm
 It may also add 0.4ms off after every 3 on/off cycles

 ---
 Este correo electrónico y, en su caso, cualquier fichero anexo al mismo, contiene 
 información de carácter confidencial exclusivamente dirigida a su destinatario o 
 destinatarios. Queda prohibida su divulgación, copia o distribución a terceros sin 
 la previa autorización escrita de Indra. En el caso de haber recibido este correo 
 electrónico por error, se ruega notificar inmediatamente esta circunstancia mediante 
 reenvío a la dirección electrónica del remitente.

 The information in this e-mail and in any attachments is confidential and solely for 
 the attention and use of the named addressee(s). You are hereby notified that any 
 dissemination, distribution or copy of this communication is prohibited without the 
 prior written consent of Indra. If you have received this communication in error, 
 please, notify the sender by reply e-mail
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Re: [Asterisk-Users] cisco ATA186 I2 vs I1

2003-09-05 Thread Michael Graff

Samy Touati [EMAIL PROTECTED] writes:

Will the I2 version work in Canada with regular anlog phones, or will
I need to change it.

No idea...  I'm not certain what Canada does with analog phones.  I
suspect they're the same as the US ones.

--Michael
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[Asterisk-Users] Ericsson webswitch 100 G4 and Asterisk

2003-09-05 Thread Senad Jordanovic

Hi,

Just got hold of Ericsson webswitch 100 G4 (4 FXO ports).
IT uses H323 as codec. The plan is to use it for incoming/outgoing calls on
two PSTN lines.
I have ATA 186 which is using SIP to use asterisk services.

I can not figure out:

1. where in asterisk do I edit conf files so it uses webswitch for
incoming/outgoing calls
2. how come there is no username+password on webswitch. (apparently just an
IP and port number)
3. is there any docs on gateway/gatekeeper for using H323 with asterisk.

If some could please explain this to me, I would be very grateful.

Thanks

Senad

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RE: [Asterisk-Users] Asterisk and Cisco 7960

2003-09-05 Thread Senad Jordanovic

hi, what does tr means at the end of line?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Gillham
Sent: 05 September 2003 06:29
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960

Andrew Joakimsen wrote:

exten = 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr)

This didn't work - what does the @1000 indicate?

It shouldn't be there, If it's defined as 1000 in sip.conf make your
dial string

exten = 1000,1,Dial(SIP/1000,20,Ttr)

You need 'SIP/[EMAIL PROTECTED]' if you want to tell the Cisco what line you are 
calling!

This just says I am calling the line configured as '1000' on the Cisco 
device that is defined as [1000] in sip.conf.

-Andrew

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[Asterisk-Users] app_queue input needed...

A friend and I have recently added the ability to announce the callers
position in the call queue every x seconds.. or even just inject an
anouncement every x seconds.  All setup in queues.conf and can be setup
per queue.

My next project is to add the ability to announce the callers estimated
wait time.  I want some feedback to see whats the best method to calculate
that?  What do you want just minutes? or minutes and seconds?  Or the
option to use one or the other?

I'm thinking (totaltime / totalcalls) - (now - qe-start) = current
estimated wait time.  Which would update after each call is hungup.

http://bugs.digium.com/bug_view_page.php?bug_id=214


Please let me know what you would like to see!?!?!

Thanks,
bkw
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Re: [Asterisk-Users] app_queue input needed...

2003-09-05 Thread Eric Wieling

If I was calling I would like to know either how long the the person
that's been in the queue the longest has been waiting OR an average of
how long the callers were in the queue before they were answered (over
the last X (where x in a config option) mins

On Fri, 2003-09-05 at 14:05, Brian West wrote:
 A friend and I have recently added the ability to announce the callers
 position in the call queue every x seconds.. or even just inject an
 anouncement every x seconds.  All setup in queues.conf and can be setup
 per queue.
 
 My next project is to add the ability to announce the callers estimated
 wait time.  I want some feedback to see whats the best method to calculate
 that?  What do you want just minutes? or minutes and seconds?  Or the
 option to use one or the other?
 
 I'm thinking (totaltime / totalcalls) - (now - qe-start) = current
 estimated wait time.  Which would update after each call is hungup.
 
 http://bugs.digium.com/bug_view_page.php?bug_id=214
 
 
 Please let me know what you would like to see!?!?!
 
 Thanks,
 bkw
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[Asterisk-Users] CDR not recording SIP username

In reading the source for the CDR_CSV module, I understand that it should 
use the SIP username as the account code for calls made from SIP devices. 
However, nothing is being recorded in the csv file for that field (i.e. 
blank value). Is there any way to add an account code for SIP users? I can 
always identify the SIP user from the channel identifier, but it would be 
cleaner to use an account code.

Thanks,
--Ernest
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Re: [Asterisk-Users] app_queue input needed...

On Fri, 2003-09-05 at 14:05, Brian West wrote:
 A friend and I have recently added the ability to announce the callers
 position in the call queue every x seconds.. or even just inject an
 anouncement every x seconds.  All setup in queues.conf and can be setup
 per queue.
 
 My next project is to add the ability to announce the callers estimated
 wait time.  I want some feedback to see whats the best method to calculate
 that?  What do you want just minutes? or minutes and seconds?  Or the
 option to use one or the other?
 
 I'm thinking (totaltime / totalcalls) - (now - qe-start) = current
 estimated wait time.  Which would update after each call is hungup.


I do not use queues, so accept my comments as only an opinion of how I
would like to experience them if I where a person in a queue.

Your wait time is not very accurate unless you have sufficiently large
enough pools of people to service them to offset little abnormalities.
So I would say it would be good to define an acceptable list of
announcements, then round up to the first available announcement and
play from there. Have to look up be something like...
under 3 minutes
under 5 minutes
under 8 minutes
under 10 minutes
under 15 minutes
under 20 minutes
If I ever heard a time over 20 minutes I'd hang up and call back later,
or stop doing business with the company. This limits down your number of
prompts and lowers the expectation of wait time accuracy.

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RE: [Asterisk-Users] Asterisk and Cisco 7960

On Fri, 2003-09-05 at 14:00, Senad Jordanovic wrote:
 hi, what does tr means at the end of line?

There is documentation, it is even within quick access.

From issueing a show application dial at a asterisk cli prompt I see
the following.

The option string may contain zero or more of the following characters:
  't' -- allow the called user transfer the calling user
  'T' -- to allow the calling user to transfer the call.
  'r' -- indicate ringing to the calling party, pass no audio until answered.
  'm' -- provide hold music to the calling party until answered.
  'd' -- data-quality (modem) call (minimum delay).
  'c' -- clear-channel data call (PRI-PRI only).
  'H' -- allow caller to hang up by hitting *.
  'C' -- reset call detail record for this call.
  'P[(x)]' -- privacy mode, using 'x' as database if provided.
  In addition to transferring the call, a call may be parked and then picked
up by another user.
  The optionnal URL will be sent to the called party if the channel supports
it.


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Andrew
 Gillham
 Sent: 05 September 2003 06:29
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960
 
 
 Andrew Joakimsen wrote:
 
 exten = 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
   
 
 This didn't work - what does the @1000 indicate?
 
 
 
 
 It shouldn't be there, If it's defined as 1000 in sip.conf make your
 dial string
 
 exten = 1000,1,Dial(SIP/1000,20,Ttr)
 
 
 You need 'SIP/[EMAIL PROTECTED]' if you want to tell the Cisco what line you are 
 calling!
 
 This just says I am calling the line configured as '1000' on the Cisco 
 device that is defined as [1000] in sip.conf.
 
 -Andrew
 
 
 
 
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-- 
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Re: [Asterisk-Users] CDR not recording SIP username

At 12:21 PM 9/5/2003 -0700, you wrote:
In reading the source for the CDR_CSV module, I understand that it should
use the SIP username as the account code for calls made from SIP devices.
However, nothing is being recorded in the csv file for that field (i.e.
blank value). Is there any way to add an account code for SIP users? I can
always identify the SIP user from the channel identifier, but it would be
cleaner to use an account code.
Hah! Undocumented (at least in the documentation I have) feature is that 
you can use the accountcode statement in sip.conf. Cool.

Thanks anyway,
--Ernest 

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Re: [Asterisk-Users] app_queue input needed...

 under 20 minutes
 If I ever heard a time over 20 minutes I'd hang up and call back later,
 or stop doing business with the company. This limits down your number of
 prompts and lowers the expectation of wait time accuracy.

Sprint PCS comes to mind on that longer than 20 min hold times! :P

bkw
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Re: [Asterisk-Users] The sounds of silence: silent soundfiles available

On Fri, 2003-09-05 at 01:40, John Todd wrote:

 [nonpedanticexample]
 exten = s,1,DigitTimeout(5)
 exten = s,2,ResponseTimeout(20)
 exten = s,3,Background(type-your-selection)
 exten = s,4,Background(silence/3)
 exten = s,5,Background(type-your-selection)
 exten = s,6,Background(silence/3)
 exten = s,7,Background(if-you-need-help-press-pound-for-instructions)
 exten = t,1,Goto(s,3)


This is another way. Please think about limiting your loops like I have
below. It is possible to get a channel that didn't detect a hangup, and
would stay busy the way you have listed above.
[anotherway]
exten = s,1,SetVar(Loop=0)
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(20)
exten = s,4,Background(type-your-selection)
exten = t,1,SetVar(Loop=[${Loop}+1])
exten = t,2,gotoif([${Loop}  6]?t|100)
exten = t,3,gotoif([${Loop} == 3]?t|200)
exten = t,4,goto(s|3)
exten = t,100,Hangup

exten = t,200,Background(if-you-need-help-press-pound-for-instructions)
exten = t,201,goto(s|3)


--
Steven Critchfield [EMAIL PROTECTED]
As usual, there is more than one way to skin a Aster-cat.

My example was intentionally incomplete as a whole dialplan, as it 
was an example.  I normally use AbsoluteTimeout for any calls coming 
into the system to handle stuck callers, and then change the 
AbsoluteTimeout to a different value when I send the caller to a Dial 
routine.  This method reduces complexity, and allows me to jump 
around in the dialplan without having to build a bunch of GotoIf 
jumps or think about where I am in a counter routine when I move 
between contexts.  Of course, the incrementing counter may be used 
for other purposes, so it may be suitable in certain circumstances.

While I really like GotoIf and use it extensively, I really hate it 
as well, since every time I make a change in a matching group, I have 
to increment every number after it, and then find every single GotoIf 
that references priorities that have been changed.

JT
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Re: [Asterisk-Users] app_queue input needed...

On Fri, 2003-09-05 at 14:41, Brian West wrote:
  under 20 minutes
  If I ever heard a time over 20 minutes I'd hang up and call back later,
  or stop doing business with the company. This limits down your number of
  prompts and lowers the expectation of wait time accuracy.
 
 Sprint PCS comes to mind on that longer than 20 min hold times! :P

I guess thats why I use a cricket phone. No stupid billing crap to argue
about, just pay before due date or be cut off.
-- 
Steven Critchfield  [EMAIL PROTECTED]

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RE: [Asterisk-Users] app_queue input needed...

2003-09-05 Thread McAughan, Matt

Title: RE: [Asterisk-Users] app_queue input needed...





There is one thing you have to look out for. Wait time is affected only by the number of calls in front of you, not total calls, the number of agents answering, and the length of calls.

I say this because if you are going to update the announcer x seconds, depending on the calculation the caller may experience, your wait time is 5 minutes, your wait time is 25 minutes, your wait time is 7 minutes. That makes me want to hang up as well. On our (non Asterisk) phone system we avoid this by just announcing the average wait time once, when the caller enters the queue.

I think the proper calculation should be a running average of time to answer over the last X period, with a factor taking in to account average agents logged in over that period. Something like average wait time per agent per period. Then factor that against your current queue position (calls entering behind you have no affect on your wait time) and current number of agents (more could log in to help out or drop off). What the proper period factor is I don't know. 30 seconds, 1 minute, after each hang up? Too frequently it will fluctuate wildly. Too infrequent it will show residual affects.

-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]]
Sent: Friday, September 05, 2003 02:25
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] app_queue input needed...



On Fri, 2003-09-05 at 14:05, Brian West wrote:
 A friend and I have recently added the ability to announce the callers
 position in the call queue every x seconds.. or even just inject an
 anouncement every x seconds. All setup in queues.conf and can be setup
 per queue.
 
 My next project is to add the ability to announce the callers estimated
 wait time. I want some feedback to see whats the best method to calculate
 that? What do you want just minutes? or minutes and seconds? Or the
 option to use one or the other?
 
 I'm thinking (totaltime / totalcalls) - (now - qe-start) = current
 estimated wait time. Which would update after each call is hungup.



I do not use queues, so accept my comments as only an opinion of how I
would like to experience them if I where a person in a queue.


Your wait time is not very accurate unless you have sufficiently large
enough pools of people to service them to offset little abnormalities.
So I would say it would be good to define an acceptable list of
announcements, then round up to the first available announcement and
play from there. Have to look up be something like...
under 3 minutes
under 5 minutes
under 8 minutes
under 10 minutes
under 15 minutes
under 20 minutes
If I ever heard a time over 20 minutes I'd hang up and call back later,
or stop doing business with the company. This limits down your number of
prompts and lowers the expectation of wait time accuracy.


-- 
Steven Critchfield [EMAIL PROTECTED]


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RE: [Asterisk-Users] Asterisk and Cisco 7960

2003-09-05 Thread Senad Jordanovic

thanks very much...

do you know of any other links to documentation, guides, manuals etc.
(Digium site
does not offer much).
The biggest problem so far, I find is lack of docs.
To produce information one does need data.


Senad


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: 05 September 2003 20:27
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk and Cisco 7960


On Fri, 2003-09-05 at 14:00, Senad Jordanovic wrote:
 hi, what does tr means at the end of line?

There is documentation, it is even within quick access.

From issueing a show application dial at a asterisk cli prompt I see
the following.

The option string may contain zero or more of the following characters:
  't' -- allow the called user transfer the calling user
  'T' -- to allow the calling user to transfer the call.
  'r' -- indicate ringing to the calling party, pass no audio until
answered.
  'm' -- provide hold music to the calling party until answered.
  'd' -- data-quality (modem) call (minimum delay).
  'c' -- clear-channel data call (PRI-PRI only).
  'H' -- allow caller to hang up by hitting *.
  'C' -- reset call detail record for this call.
  'P[(x)]' -- privacy mode, using 'x' as database if provided.
  In addition to transferring the call, a call may be parked and then picked
up by another user.
  The optionnal URL will be sent to the called party if the channel supports
it.


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Andrew
 Gillham
 Sent: 05 September 2003 06:29
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960


 Andrew Joakimsen wrote:

 exten = 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
 
 
 This didn't work - what does the @1000 indicate?
 
 
 
 
 It shouldn't be there, If it's defined as 1000 in sip.conf make your
 dial string
 
 exten = 1000,1,Dial(SIP/1000,20,Ttr)
 

 You need 'SIP/[EMAIL PROTECTED]' if you want to tell the Cisco what line you are
 calling!

 This just says I am calling the line configured as '1000' on the Cisco
 device that is defined as [1000] in sip.conf.

 -Andrew




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Re: [Asterisk-Users] The sounds of silence: silent soundfiles available

On Fri, 2003-09-05 at 14:41, John Todd wrote:
 On Fri, 2003-09-05 at 01:40, John Todd wrote:
 
   [nonpedanticexample]
   exten = s,1,DigitTimeout(5)
   exten = s,2,ResponseTimeout(20)
   exten = s,3,Background(type-your-selection)
   exten = s,4,Background(silence/3)
   exten = s,5,Background(type-your-selection)
   exten = s,6,Background(silence/3)
   exten = s,7,Background(if-you-need-help-press-pound-for-instructions)
 
   exten = t,1,Goto(s,3)
 
 
 This is another way. Please think about limiting your loops like I have
 below. It is possible to get a channel that didn't detect a hangup, and
 would stay busy the way you have listed above.
 
 [anotherway]
 exten = s,1,SetVar(Loop=0)
 exten = s,2,DigitTimeout(5)
 exten = s,3,ResponseTimeout(20)
 exten = s,4,Background(type-your-selection)
 
 
 exten = t,1,SetVar(Loop=[${Loop}+1])
 exten = t,2,gotoif([${Loop}  6]?t|100)
 exten = t,3,gotoif([${Loop} == 3]?t|200)
 exten = t,4,goto(s|3)
 
 exten = t,100,Hangup
 
 exten = t,200,Background(if-you-need-help-press-pound-for-instructions)
 exten = t,201,goto(s|3)
 
 
 
 --
 Steven Critchfield [EMAIL PROTECTED]
 
 As usual, there is more than one way to skin a Aster-cat.
 
 My example was intentionally incomplete as a whole dialplan, as it 
 was an example.  I normally use AbsoluteTimeout for any calls coming 
 into the system to handle stuck callers, and then change the 
 AbsoluteTimeout to a different value when I send the caller to a Dial 
 routine.  This method reduces complexity, and allows me to jump 
 around in the dialplan without having to build a bunch of GotoIf 
 jumps or think about where I am in a counter routine when I move 
 between contexts.  Of course, the incrementing counter may be used 
 for other purposes, so it may be suitable in certain circumstances.

While that is another way of doing it, it would seem odd to have a
prompt be terminated mid sentence/word. Of course if set appropriately
long enough, the pbx operator wouldn't care about the user that was cut
off as they aren't being productive. 

Just a request, but if you post examples like this, please make it
complete enough to show best practices. I'm sure someone before too long
is going to scour the archive and create a book on this. Maybe even an
O'reilly book if we are lucky. It would be nice to make sure everyone
understands these concepts that are often glazed over in interest of
brevity or reduction of complexity.  

 While I really like GotoIf and use it extensively, I really hate it 
 as well, since every time I make a change in a matching group, I have 
 to increment every number after it, and then find every single GotoIf 
 that references priorities that have been changed.

I hear you, and agree. 
-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] app_queue input needed...



If I was calling I would like to know either how long the the person
that's been in the queue the longest has been waiting OR an average of
how long the callers were in the queue before they were answered (over
the last X (where x in a config option) mins
On Fri, 2003-09-05 at 14:05, Brian West wrote:
 A friend and I have recently added the ability to announce the callers
 position in the call queue every x seconds.. or even just inject an
 anouncement every x seconds.  All setup in queues.conf and can be setup
 per queue.
 My next project is to add the ability to announce the callers estimated
 wait time.  I want some feedback to see whats the best method to calculate
 that?  What do you want just minutes? or minutes and seconds?  Or the
 option to use one or the other?
 I'm thinking (totaltime / totalcalls) - (now - qe-start) = current
 estimated wait time.  Which would update after each call is hungup.
 http://bugs.digium.com/bug_view_page.php?bug_id=214

 Please let me know what you would like to see!?!?!

 Thanks,
  bkw

--
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)


Like others, I rarely use queues and so my notes below are based only 
on what I would like to experience as a queue member.

I would tend to agree with the idea of letting the caller hear the 
longest wait time when they first get an answer, and it would 
certainly be easy to announce the time that the longest call has been 
waiting in the queue instead of doing some sort of calculation based 
on averages, etc.   However, that announcement would only be useful 
as the first announcement made, since that caller would then progress 
through the queue and their wait time would be reduced as they get 
closer to being the next person picked up.  So, the longest wait 
time is only useful as a first announcement.

The second method, where a sliding window average of wait times in 
the last X minutes is used as the sample base is a bit more 
difficult, but after some thought I am think it will provide a more 
accurate number.  Note that an unanticipated result of this method 
may be that some callers hear their queue wait time increase instead 
of decrease, which may have unpredictable results on customers. :-)

JT
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RE: [Asterisk-Users] app_queue input needed...

So announce this when they first enter once and only once.  That
sounds like a reasonable idea.

bkw

On Fri, 5 Sep 2003, McAughan, Matt wrote:

 There is one thing you have to look out for. Wait time is affected only by
 the number of calls in front of you, not total calls, the number of agents
 answering, and the length of calls.

 I say this because if you are going to update the announcer x seconds,
 depending on the calculation the caller may experience, your wait time is 5
 minutes, your wait time is 25 minutes, your wait time is 7 minutes.
 That makes me want to hang up as well. On our (non Asterisk) phone system we
 avoid this by just announcing the average wait time once, when the caller
 enters the queue.

 I think the proper calculation should be a running average of time to answer
 over the last X period, with a factor taking in to account average agents
 logged in over that period. Something like average wait time per agent per
 period. Then factor  that against your current queue position (calls
 entering behind you have no affect on your wait time) and current number of
 agents (more could log in to help out or drop off). What the proper period
 factor is I don't know. 30 seconds, 1 minute, after each hang up? Too
 frequently it will fluctuate wildly. Too infrequent it will show residual
 affects.

 -Original Message-
 From: Steven Critchfield [mailto:[EMAIL PROTECTED]
 Sent: Friday, September 05, 2003 02:25
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] app_queue input needed...


 On Fri, 2003-09-05 at 14:05, Brian West wrote:
  A friend and I have recently added the ability to announce the callers
  position in the call queue every x seconds.. or even just inject an
  anouncement every x seconds.  All setup in queues.conf and can be setup
  per queue.
 
  My next project is to add the ability to announce the callers estimated
  wait time.  I want some feedback to see whats the best method to calculate
  that?  What do you want just minutes? or minutes and seconds?  Or the
  option to use one or the other?
 
  I'm thinking (totaltime / totalcalls) - (now - qe-start) = current
  estimated wait time.  Which would update after each call is hungup.


 I do not use queues, so accept my comments as only an opinion of how I
 would like to experience them if I where a person in a queue.

 Your wait time is not very accurate unless you have sufficiently large
 enough pools of people to service them to offset little abnormalities.
 So I would say it would be good to define an acceptable list of
 announcements, then round up to the first available announcement and
 play from there. Have to look up be something like...
 under 3 minutes
 under 5 minutes
 under 8 minutes
 under 10 minutes
 under 15 minutes
 under 20 minutes
 If I ever heard a time over 20 minutes I'd hang up and call back later,
 or stop doing business with the company. This limits down your number of
 prompts and lowers the expectation of wait time accuracy.

 --
 Steven Critchfield  [EMAIL PROTECTED]

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RE: [Asterisk-Users] Hardware IAX phone (please read and reply!)

2003-09-05 Thread John Laur


 Steven Critchfield wrote:
 
  On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote:
 This is a completely open-source and open-hardware hardware phone
based
 on Linux on an ARM embedded platform ... they already had lots of
 experience ... but might need some different software ...
 
  bzzzt. wrong. There is a lot known about the hardware but it is not
  open. The software is only open after it was reloaded with debian.
Also
  while the site you list was cheap, if you dig round, the
manufacturing
  cost was over $300 each and target retail was over %600. Granted
that
  was over 3 years ago, it wouldn't have dropped in price too
  significantly. The site you list was liquidating the last known
  inventory of those units.

The other problem with the touchscreens and VoIP is that the telephone
audio circuitry was not accessible by software running on the phone.
Here is a block diagram:

http://www.blurbco.com/~gork/tuxscreen/shanblock.gif

A modification (ShanIP2) was designed to make the handset/speakerphone
audio to/from the dsp accessible via the UCB1200 audio chip, and I had
designed a PCB for the circuit here:

http://www.blurbco.com/~gork/tuxscreen/shanip2-gork8.gif

 So have a look there :
 
 http://www.lart.tudelft.nl/
 
 You will find there the hardware that evolved from what was in the
 Tuxscreen. It's license is open. It runs a 220Mhz StrongARM with more
 than 200 MIPS and has options for ethernet and sound i/o, all is
 linux-compatible ...

The LART was actually around before the tuxscreen, and although it is
similar, you'll find that most SA1XXX based designs are. It is still a
good little board and fun to work with, as is the Tuxscreen if you can
still pick one up used from someone. Anyway, since this is starting now
getting pretty offtopic, I should probably leave it at this...

John

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[Asterisk-Users] Re: Asterisk Jitters

2003-09-05 Thread Zak

Message: 2
Subject: Re: [Asterisk-Users] Re: Asterisk Jitters
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Fri, 05 Sep 2003 11:54:19 -0500
Reply-To: [EMAIL PROTECTED]
As you can see wcfxo is still sharing an IRQ.  It won't work well if it
shares an IRQ.
I have changed the pci slot of the fxo so that it won't share IRQ with 
another device but the jittering is still
there.  check the interrupts list below

  CPU0
 0:1158022  XT-PIC  timer
 1:807  XT-PIC  keyboard
 2:  0  XT-PIC  cascade
 5:1377109  XT-PIC  eth0, Intel ICH2
 8:  1  XT-PIC  rtc
10:   12231674  XT-PIC  wcfxo
12:  14239  XT-PIC  PS/2 Mouse
14:  18691  XT-PIC  ide0
15:  21804  XT-PIC  ide1
NMI:  0
ERR:  0
On Fri, 2003-09-05 at 19:39, Zak wrote:

Hi Steven.

I have done as you suggested and I'm still getting the same problem.
/proc/interrupts lists the following:
0:  45489  XT-PIC  timer
 1:235  XT-PIC  keyboard
 2:  0  XT-PIC  cascade
 5: 335816  XT-PIC  wcfxo, Intel ICH2
 8:  1  XT-PIC  rtc
 9:  0  XT-PIC  usb-uhci
10:829  XT-PIC  eth0
11:  0  XT-PIC  usb-uhci
12:194  XT-PIC  PS/2 Mouse
14:   4402  XT-PIC  ide0
15:  2  XT-PIC  ide1
NMI:  0
ERR:  0
I am also getting the following message when asterisk starts.. but I'm
not sure if it means anything?
WARNING[16384]: File chan_oss.c, Line 974 (load_module): XXX I don't
work right
with non-full
duplex sound cards XXX
 == Registered channel type 'Console' (OSS Console Channel Driver)
 == Parsing '/etc/asterisk/oss.conf': Found
WARNING[114696]: File chan_oss.c, Line 232 (sound_thread): Read error
on sound
device: Resource temporarily unavailabl
thank,

Zak

Bing,Bing,Bing, we have the problem. nvidia and wcfxo cards on the same
interupt.

I'd say try removing a 2 WCFXO cards from the system and see if the
interupts free up, and your jitter stops.

12: 524504  XT-PIC  PS/2 Mouse
14: 165140  XT-PIC  ide0
15: 281208  XT-PIC  ide1
NMI:  0
ERR:  0

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[Asterisk-Users] Asterisk phone system plan - for review!

2003-09-05 Thread Mike Ciholas


Hi all,

I would be most grateful if someone would review my plans for my
new phone system and comment on areas of expected trouble and
advice on what to do better.  Instead of moving our Panasonic
KX-TD1232/TVS200 system (ugh...) to our new location, we've
decided to jump into IP telephony with *.  But we are new to *
(but not Linux), so we're trying to learn as much as we can
before we jump in and drown.  I've got 6 weeks to make this work.

The basic plan is as follows:

1. A T1 for local phone service has 8 live channels (and 16
dead ones) and uses robbed bit signaling (that is, not a PRI
which was *way* more expensive, why is that?).  A local CLEC
(Cinergy Communications)  can provide the T1 for $75/mo plus
$15/active channel.  I believe this comes with CID w/name plus a
bunch of other features too long to list but probably typical for
this service.  Anything to watch out for in getting a T1?  
Local, long distance, and toll free inbound are provided over the
T1 (5.9c/min for LD and toll free).  I'm not quite ready to jump
to IP dial tone service at this time but I am keeping an eye on
that.

2. The T1 will be terminated on one port of a Digium TE410P, 
$1500.  There seems to be no question this is the right T1 
interface to buy.  There isn't a single Digium product for sale 
on Ebay.  Guess everybody wants to keep theirs!

3. A rack mount PC running * will be in the server room.  It
will have two ethernet interfaces, one for the phones and one for
the internal LAN/internet.  The phones will be on a subnet with
the * box doing the routing between the PC LAN and phone LAN
(which should not be much traffic, right?).  The PC will also
host the voicemail stuff (which I have yet to investigate).  I
have not selected PC hardware in detail, any suggestions?  
Looking at ~$1000 for the PC.

4. The phone LAN will be served by a dedicated ethernet switch.  
I'd love to get one with inline power (Cisco C3524-PWR for
example) but these are expensive relative to garden variety
switches.  We may end up using commodity switches and wall bricks
at each phone until PoE switches become commodity (which they
most certainly will become in 1-2 years).  Do I bite the bullet
and buy a C3524-PWR ($1600 Ebay), do an inline power hub ($?), or
wall bricks ($40/phone?)?  Part of me wants to just take a 48
volt power supply and do some hack in the wiring closet...

5. The desk phones are likely to be Cisco 7940/60 series and run 
SIP.  The phones will have their own LAN jack separate from PC 
traffic.  We are looking at about 25 phones total.

6. For legacy analog phones (fax machine, certain cordless
phones, etc), I will buy an Adtran TA 750 channel bank with FXS
cards (perhaps as many as 24).  Going price seems to be around
$400 for this.  I can add a few FXO ports if I want some POTS
backup lines.  The Adtran would tie into a port of the TE410P.

7. For dial in service, I will buy a Lucent Portmaster 3.  While
this is *serious* overkill for my needs (which are met with 2
analog modems now), it is cheap enough (~$200 Ebay) and it
provides direct to digital modems.  I should be able to get
reasonable connect speeds, perhaps up to 53K (is this true with
T1 robbed bit signaling?).  The PM3 will be tied to one port of
the TE410P Digium card.  If I am feeling gratuitously silly, I
can connect a second T1 to the PM3 and have 48 modems available
(on 8 incoming lines...?).  I know dial in modem service is going
the way of the dodo bird, but having a means for traveling
employees to dial a toll free number and get to the internet is
still very handy!  Presently, our analog modems are passed
through the Panasonic switch which cuts our connect performance
to about 24K (bleah...).  I cannot dedicate an incoming line to 
the modems since they are used sporadically.

8. The power infrastructure is to concentrate as much of the
equipment in the server room and provide everything with UPS
backed power.  Theoretically, during a power failure, all of the
systems will work until the UPS has been depleted.  This assumes
the phones are powered over the LAN cable.  This means we should
have phone service and internet service during power outage.  
UPS recommendations?  I'd like at least 1 hour, and 3 would be
nice!

9. From the software setup, given that the incoming T1 will be
G.711 ulaw (right?), I would probably force everything internal
to be ulaw so the * box has no codec work to do, just shuffling
bits.  I'm not concerned about the bandwidth used on the internal
phone LAN.  This means the TE410P card just shuffles around bytes
between ports/channels (for T1 to channel bank or modems) and *
shuffles bytes between T1 and phone LAN.  Sounds like this is the
simplest setup.  Of course, I don't know what echo canceling will
do to this.  I would guess we would want to hard code that all
calls to/from the modem get no echo canceling.

10. On an experimental basis, it might be nice for certain
employees to be able to have an IP phone at their house

Re: [Asterisk-Users] app_queue input needed...

 The second method, where a sliding window average of wait times in
 the last X minutes is used as the sample base is a bit more
 difficult, but after some thought I am think it will provide a more
 accurate number.  Note that an unanticipated result of this method
 may be that some callers hear their queue wait time increase instead
 of decrease, which may have unpredictable results on customers. :-)

I think when they first enter the queue will be fine and even better than
trying to recalc the callers estimated wait time.

bkw

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[Asterisk-Users] Noisy/Clicky hangup

2003-09-05 Thread Matt Lawson

When I call in from an outside POTS line to a Zap channel, and the call 
ends, it seems like the hangups are very sloppy.  I see Asterisk give 
the hangup command, but on my phone there's lots of clicks and the line 
acts like it's staying open for several seconds, then I hear a phone 
ringing sound followed by If you'd like to make a call, please hang up 
and try again...

Is there something wrong with my setup that it acts this way, or is that 
just how it is?

Thanks

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[Asterisk-Users] chan_zap Cannot handle frames in 2 format

2003-09-05 Thread Matt Lawson

I have discovered something quirky in our Asterisk.  If I call in to a 
Zap channel (from an outside POTS line), then transfer the call around 
several times, I get the above error, after which it will hangup.  I 
believe Asterisk may issue a SIP CANCEL to the extension it was starting 
to dial.

Now when I say 'transferred around several times,' our routing is pretty 
compex and uses the database lookup for user extensions.  It plays a 
static message, then goes to a 'which user do you want' type menu, then 
may go to voicemail or ring an extension, while I beat on it with 
Redirect commands through the management interface.  I sometimes 
redirect it to specific SIP extensions and sometimes to users, which 
have to be looked up in the database.

The SIP phones are set to communicate with Asterisk only using mu-law. 
The IAX connection uses GSM (and we do have multiple Asterisks talking 
over IAX).  One thing that puzzles me is that the format 2 in chan_zap 
I believe corresponds to GSM.  Where is it getting that?  It should only 
be using mu-law on the local system.

The only other possibility that occurs to me, is that voicemails are 
left in GSM format.  Is it possible that if a call gets transferred 
after it's already in the process of leaving a voicemail that will break 
it?

Suggestions?

Thanks,

Matt

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[Asterisk-Users] T1 - A little guidance needed to get started, What order to do zaptel, zapata...

2003-09-05 Thread mvickers


I have about a dozen SIP phones up and working, now I want to connect the
asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers
conencted to the Fujitsu PBX that I built with mgetty/pppd and have the
lines provisioned the same way as those dial-up server, ESF, B8ZS, and EM
wink start, so I have confidence in the guys who set up the PBX.

I've built a loop back plug for T1 and looped it both directions, I can
get the Digum T100P card to go green as well as the PBX port. So I have
confidence in the wiring.

I don't understand the relation ship between zaptel and zapata and wether
I need to config a dail plan or anything else in asterisk before I get
the T1 up. So do I start with /etc/zaptel.conf and the zaptel module.

Then work with the zapata stuff (Does this have a module), then asterisk.

There seems to be a lot of docs scattered around about asterisk is a good
list where docs are, and some kind of overview.  I'm a router/linux guy
not a PBX guy and the architecture and nomenclature are unfamilure to me,
is there a PBX book I should read for background info.

Oh and how does one make the equivalent of a defaut route in the dial
plan, ie any call not listed in the dial plan goes out the T1

Thanks in advance.


Mark Vickers, RealNetworks Inc.  Desk: (206) 674-2391  Fax: (206)674-3588
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[Asterisk-Users] Moh

2003-09-05 Thread Ben Bloomberg

Would anyone mind emailing me, or maybe posting somewhere their music 
on hold .so file?

thx

-ben

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[Asterisk-Users] SIP Phone to use with *

2003-09-05 Thread Langley, Sean

Any recommendations on a hardware based SIP phone to use with *?

I'm looking for something that would be common, as well as quick and easy to source, 
somthing relatively quick and easy to configure.

Side note, is SIP automatically enabled in *, or do I have to add a channel driver as 
I do with H323?

Regards,

Sean Langley, P.Eng

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Re: [Asterisk-Users] SIP Phone to use with *

2003-09-05 Thread Rich Adamson

Sean,

 Any recommendations on a hardware based SIP phone to use with *?
 
 I'm looking for something that would be common, as well as quick and easy 
 to source, somthing relatively quick and easy to configure.

I'm very new to this as well, but with 20+ years of telephony and data
network performance background.

Between the Snom 200 and Cisco 7960 (both seem to work well), the Snom
is much quicker to deploy, you don't need a tftp server (which is pretty
much a requirement for the 7960), takes up less desktop space, etc, etc.
Both phones provide roughly the same functionality, however if you have
folks that like lots of buttons or like well-recognized names, the Cisco
does a fine job.

From what I've seen on the list, there are many other choices as well, I
just don't have any experience/knowledge of those.



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RE: [Asterisk-Users] Noisy/Clicky hangup

2003-09-05 Thread Adam Roach

This is an oddity of how the POTS works, and has nothing
to do with asterisk.

For almost all domestic switches in the world, the called
party can hang up the handset without disconnecting
the call. If the phone is picked up before a timer
pops (on the order of 10-30 seconds), then the call
continues. Only after this timer expires does the call
actually get disconnected. Ostensibly, this gives
the called party (who may not be prepared to receive
a call) the opportunity to hang up, move (briskly)
to another phone, and pick back up. In practice, it's
silly because so few people know it works that way.

The clicks and pops are just line noise. You'll probably
hear the same thing if you pick up your phone, hit a
single touch tone (not 0), and then listen.

/a

 -Original Message-
 From: Matt Lawson [mailto:[EMAIL PROTECTED]
 Sent: Friday, September 05, 2003 16:08
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Noisy/Clicky hangup
 
 
 When I call in from an outside POTS line to a Zap channel, 
 and the call 
 ends, it seems like the hangups are very sloppy.  I see 
 Asterisk give 
 the hangup command, but on my phone there's lots of clicks 
 and the line 
 acts like it's staying open for several seconds, then I hear a phone 
 ringing sound followed by If you'd like to make a call, 
 please hang up 
 and try again...
 
 Is there something wrong with my setup that it acts this way, 
 or is that 
 just how it is?
 
 Thanks
 
 
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[Asterisk-Users] VONAGE or IP Dialtone

2003-09-05 Thread asterisk









The
Vonage service is offered with a SIP Cisco ATA device for connection to an
analog phone.



Is
it possible to connect the Vonage service directly to the Asterisk PBX
bypassing the ATA and FXO card? Are
there other services that offer this capability or something similar to IP
dialtone?



Thanks



Kevin

RE: [Asterisk-Users] VONAGE or IP Dialtone

2003-09-05 Thread Andrew Joakimsen









No. You can use packet8 if you slightly
modify the asterisk source code (outgoing calls only) or you can use the
service provided by nufone.net





-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Friday, September 05, 2003
7:21 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] VONAGE
or IP Dialtone



The Vonage service is
offered with a SIP Cisco ATA device for connection to an analog phone.



Is it possible to connect
the Vonage service directly to the Asterisk PBX bypassing the ATA and FXO
card? Are there other services that offer this capability or something
similar to IP dialtone?



Thanks



Kevin

Re: [Asterisk-Users] T1 - A little guidance needed to get started, What order to do zaptel, zapata...

2003-09-05 Thread mvickers

More info:

cat /etc/zaptel.conf |grep -v ^#
span=1,1,0,esf,b8zs
em=1-24
loadzone = us
defaultzone=us

cat /etc/asterisk/zapata.conf |grep -v ^;
[channels]
context=default
signalling=em_w
group=1
channel = 1-24

lsmod:
Module  Size  Used byNot tainted
wct1xxp12320  24
zaptel180416  50  [wct1xxp]
ppp_generic15776   0  [zaptel]
slhc4384   0  [ppp_generic]

dmesg:
Freed a Wildcard
Zapata Telephony Interface Unloaded
PPP generic driver version 2.4.2
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 9 for device 01:09.0
PCI: Sharing IRQ 9 with 00:1f.3
Framer: DS21552, Revision: 3 (T1)
Found a Wildcard: Digium Wildcard T100P T1/PRI
Registered tone zone 0 (United States / North America)
Using ESF/B8ZS coding/framing
Calling startup (flags is 4099)


Thanks again


Mark Vickers, RealNetworks Inc.  Desk: (206) 674-2391  Fax: (206)674-3588

On Fri, 5 Sep 2003 [EMAIL PROTECTED] wrote:


 I have about a dozen SIP phones up and working, now I want to connect the
 asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers
 conencted to the Fujitsu PBX that I built with mgetty/pppd and have the
 lines provisioned the same way as those dial-up server, ESF, B8ZS, and EM
 wink start, so I have confidence in the guys who set up the PBX.

 I've built a loop back plug for T1 and looped it both directions, I can
 get the Digum T100P card to go green as well as the PBX port. So I have
 confidence in the wiring.

 I don't understand the relation ship between zaptel and zapata and wether
 I need to config a dail plan or anything else in asterisk before I get
 the T1 up. So do I start with /etc/zaptel.conf and the zaptel module.

 Then work with the zapata stuff (Does this have a module), then asterisk.

 There seems to be a lot of docs scattered around about asterisk is a good
 list where docs are, and some kind of overview.  I'm a router/linux guy
 not a PBX guy and the architecture and nomenclature are unfamilure to me,
 is there a PBX book I should read for background info.

 Oh and how does one make the equivalent of a defaut route in the dial
 plan, ie any call not listed in the dial plan goes out the T1

 Thanks in advance.


 Mark Vickers, RealNetworks Inc.  Desk: (206) 674-2391  Fax: (206)674-3588
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[Asterisk-Users] SIP and NAT traversal

2003-09-05 Thread Serge Mankovski

Hi All,
i found an article that explains SIP NAT woes.
http://www.sipcenter.com/files/SIPNATtraversal.pdf

It is a great read for all people in the mailing list that have problems 
with SIP when * is behind NAT or when there is NAT between asterisk and a 
SIP phone.

Serge

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Re: [Asterisk-Users] Moh

Why on earth don't you just compile it?

bkw

On Fri, 5 Sep 2003, Ben Bloomberg wrote:

 Would anyone mind emailing me, or maybe posting somewhere their music
 on hold .so file?

 thx

 -ben

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Re: [Asterisk-Users] Moh

At 08:38 PM 9/5/2003 -0500, you wrote:
Why on earth don't you just compile it?
Thank you! I was going to ask but didn't want to look stupid :)

--Ernest


bkw

On Fri, 5 Sep 2003, Ben Bloomberg wrote:

 Would anyone mind emailing me, or maybe posting somewhere their music
 on hold .so file?

 thx
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[Asterisk-Users] SIP + NAT question

I have a few questions regarding SIP and NAT that you may be able to 
answer. In both cases, I'm assuming that the customer will use SNOM 
phones and/or xten soft-phones.

Q1: I know that it is possible to use a STUN server to handle SIP over NAT. 
Does this require any special configuration of the NAT router? For example, 
will I need to configure port forwarding?

Q2: If I know the external port of the NAT router, and if I know that it 
will never change, do I still need a STUN server? In other words, can the 
SNOM and Xten (soft)phones be configured to know the proper addresses 
without discovering them through STUN?

Thanks,
--Ernest
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Re: [Asterisk-Users] Moh

2003-09-05 Thread Ben Bloomberg

At this point, I'm just trying to save what I have. The installation 
I'm using is the one that comes with debian and it doesn't include any 
files relating to music on hold. (I'm also a complete and total newbie) 
So, if there is a way to recompile the package with music on hold, that 
would be awesome, but so far I haven't found a way to do it.

-ben

On Friday, September 5, 2003, at 09:38  PM, Brian West wrote:

Why on earth don't you just compile it?

bkw

On Fri, 5 Sep 2003, Ben Bloomberg wrote:

Would anyone mind emailing me, or maybe posting somewhere their music
on hold .so file?
thx

-ben

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[Asterisk-Users] ISDN Primary Rate Interface (PRI) - 2B Transfer

2003-09-05 Thread Kevin Fjelsted




 

Does * support ISDN 
Primary Rate Interface (PRI) - 2B Transfer Capability for 
T-1/PRI?
In other words 
the ability to take a call and join it to another call and then drop off letting 
the CO-switch take over.

-Kevin

Kevin Fjelsted, PresidentAltiCom CTI, 
Inc.

Track Me Down!One number Access, Press 11# 
during the voice mail message greetingto have me F-O-U-N-D!

Phone: 
612.259.0722Fax: 
612.259.0723VoIP: 65.209.158.245 Ext. 
222

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[Asterisk-Users] Bug in my head or bug in the code?

I am having Yet Another Regular Expression problem, but this one 
might not be my fault, or at least it might not be obviously my 
fault.  :-)

exten = 2212,1,SetVar(FOO=123456**)
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = *])
This script continues with a value of 0 in BAR.

Similarly, none of the following changes made a difference in that 
result, which is expected since the * is not listed in 
README.variables as a character that must be escaped:

exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = *])
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = \*])
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = \*])
I have also tried setting the variable ${BAZ}=*  and then using that 
in my comparison, with the same unexpected results.

Oddly enough, this almost-identical example below has different, but 
normal, results: BAR=1

exten = 2212,1,SetVar(FOO=123456##)
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = #])
What gives?  Am I colliding with a problem that is the result of the 
* character being used in expr evaluations and somehow not being 
handled correctly, or am I simply not implementing the syntax 
correctly?

JT
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Re: [Asterisk-Users] ISDN Primary Rate Interface (PRI) - 2B Transfer