Re: [Asterisk-Users] BYEXTENSION and DBPut

[Walt Davis]

Hey I found it, CALLERIDNUM ... cl!

> So using ${EXTEN} works for most things but I was trying to implement a
> whitelist as follows:
>
> exten => s,6,DBGet(WLN=whitelist/${EXTEN})
> exten => s,7,Goto(ringhouse,s,1)
>
> exten => s,107,Goto(nothome,s,1)
>
> However, in the console I see that ${EXTEN} gets replaced with 's' and
> what I want is the the CID number. The must be another variable for that
> but what is its name?
>
>> Don't use BYEXTENSION use ${EXTEN}
>>
>> bkw
>>
>> On Sat, 20 Dec 2003, Walt Davis wrote:
>>
>>> Hey I need another pair of eyes on this!
>>>
>>> I would like to add phones numbers to the blacklist from any handset
>>> so I did this:
>>>
>>>  exten => _*66XX,1,StripMSD,3
>>>  exten => _XX,2,DBPut,blacklist/BYEXTENSION/1
>>>  exten => _XX,3,Hangup
>>>
>>> However what I get in the database is:
>>>
>>>  /blacklist/BYEXTENSION : 1
>>>
>>> And BYEXTENSION is not replaced with the actual number dialed.
>>>
>>> Am I trying to do something that can not be done or am I just not
>>> doing it correctly?
>>>
>>> Walt
>>>
>>>
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>
>
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Re: [Asterisk-Users] BYEXTENSION and DBPut

[Walt Davis]

So using ${EXTEN} works for most things but I was trying to implement a
whitelist as follows:

exten => s,6,DBGet(WLN=whitelist/${EXTEN})
exten => s,7,Goto(ringhouse,s,1)

exten => s,107,Goto(nothome,s,1)

However, in the console I see that ${EXTEN} gets replaced with 's' and
what I want is the the CID number. The must be another variable for that
but what is its name?

> Don't use BYEXTENSION use ${EXTEN}
>
> bkw
>
> On Sat, 20 Dec 2003, Walt Davis wrote:
>
>> Hey I need another pair of eyes on this!
>>
>> I would like to add phones numbers to the blacklist from any handset
>> so I did this:
>>
>>  exten => _*66XX,1,StripMSD,3
>>  exten => _XX,2,DBPut,blacklist/BYEXTENSION/1
>>  exten => _XX,3,Hangup
>>
>> However what I get in the database is:
>>
>>  /blacklist/BYEXTENSION : 1
>>
>> And BYEXTENSION is not replaced with the actual number dialed.
>>
>> Am I trying to do something that can not be done or am I just not
>> doing it correctly?
>>
>> Walt
>>
>>
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Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download

[Dan]

Hi,

>- Original Message - 
>From: "González Mata David" <[EMAIL PROTECTED]>
> Hi Dan:
>
> I installed your Activex version of DIAX in Delphi 5,
> but when i try to use it in a project, i got an error
> "OLE error 800A01C5". I'm using windows 98SE.
> I hope you can help me.

This ActiveX is made to be used in a web page.
I'll check to see if I can help.

BR,
Dan


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Re: [Asterisk-Users] Dialing dead SIP peers give misleading (BUSY) voicemail result ...

[Stephen Davies]

On Sun, 21 Dec 2003, Darren Nickerson wrote:

> In the case of a physically-disconnected ZAP extension, the Dial application
> succeeds, moving on to the next step in the dialplan. That is much more in
> line with my expectation.

With an X100P card, diconnecting the card from the line results in
attempts to dial out on it also giving BUSY.  This gives similar
issues to yours.

Again, I think it should continue at the next priority rather then
branching to the +101 one.

Steve


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AW: [Asterisk-Users] 486 Busy message - SNOM 200

[Christian Stredicke]

„Everyone is busy“
is not generated by the snom phone. If you see a 486 in the trace of the phone
(maybe it’s translated into this message on Asterisk), it is generated on
the phone and then we need to guess what the reason is. 

 

Christian

 



-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von tony banks
Gesendet: Montag, 22. Dezember
2003 04:26
An:
[EMAIL PROTECTED]
Betreff: [Asterisk-Users] 486 Busy
message - SNOM 200

 

Hello
All,

Whenever I try calling SNOM 200, I am getting "Everyone is Busy at this
time" message though line is free. I tried with firmware image of
2.02t and 2.03b (this I received from SNOM customer support) but problem still
exists. Rebooting Phone doesn't help either.

So, is it a firmware problem or there might be problem with the my asterisk
configuration.

Thanks for your time.

Tony 

[Asterisk-Users] unsubscribe

[Callcenter Pulse]

Please unsubscribe the following email id - [EMAIL PROTECTED]
 
thanks
 
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Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download

[Masakazu Nakano]

I need more instruction too.

like a http://www.dairiten.com/webiax/

On Mon, 22 Dec 2003 00:02:08 -0200
Carlos Arnt <[EMAIL PROTECTED]> wrote:

> I put like the readme.txt say the code in my web page , put the OCX in the same 
> Directory, but not work.
> Did has any problem ??
> 
> Thanks
> 
> The code:
> 
> 
> 
> 
> New Page 1
> 
> 
>   CODEBASE="diax.ocx">
> 
> 
> 
> 
> 

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[Asterisk-Users] iconnect / asterisk ? calls hang up

[vocalvoip]

hi 
i got iconnect to work, works pretty well now except calls sometimes (more often than 
not) hang up after a couple of minutes.. heres a bit of the debuging

Record-Route: 
From: sip:[EMAIL PROTECTED];tag=3281050172-73809
To: "JUSTIN XLITE" ;tag=as09766a78
Call-ID: [EMAIL PROTECTED]
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


to 213.137.73.146:5060
 == Spawn extension (mpa-phones, 86189214, 2) exited non-zero on 'SIP/2001-40ae'
set_destination: Parsing  for address/port to send to
set_destination: set destination to 211.28.211.217, port 5060
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 61.95.68.84:5060;branch=z9hG4bK097c0033
From: ;tag=as240c55f5
To: justin ;tag=2602104623
Contact: 
Call-ID: [EMAIL PROTECTED]



211.28.211.217 is me at home.. exten 2001. 61.95.68.84 is server.. and the 213 is i 
connect

anyone experienced  this before.. ??

thanks heaps :)

Justin 
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[Asterisk-Users] iconnect / asterisk ? calls hang up

[vocalvoip]

hi 
i got iconnect to work, works pretty well now except calls sometimes (more often than 
not) hang up after a couple of minutes.. heres a bit of the debuging

Record-Route: 
From: sip:[EMAIL PROTECTED];tag=3281050172-73809
To: "JUSTIN XLITE" ;tag=as09766a78
Call-ID: [EMAIL PROTECTED]
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 213.137.73.146:5060
  == Spawn extension (mpa-phones, 86189214, 2) exited non-zero on 'SIP/2001-40ae'
set_destination: Parsing  for address/port to send to
set_destination: set destination to 211.28.211.217, port 5060
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 61.95.68.84:5060;branch=z9hG4bK097c0033
From: ;tag=as240c55f5
To: justin ;tag=2602104623
Contact: 
Call-ID: [EMAIL PROTECTED]



211.28.211.217 is me at home.. exten 2001. 61.95.68.84 is server.. and the 213 is i 
connect

anyone experienced  this before.. ??

thanks heaps :)

Justin
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Re: [Asterisk-Users] Queue only ringing one agent at a time

[Greg Boehnlein]

On Fri, 12 Dec 2003, Derek Barber wrote:

> but, if this is case then how can you run a call center with asterisk? 
> What if you have 40 simultaneous calls coming into the call center, most
> calls would be missed, even if you have 40 available agents.  Of course
> one call should go to one agent, but if a second call, or a third call
> joins the queue, shouldn't they also go to agents as well?  

Is it possible that the phone can handle multiple calls, so that both 
callers are actually ringing on the first phone? 





> 
> derek
> 
> On Fri, 2003-12-12 at 13:33, Brian West wrote:
> > Its going to try one at a time till its answered thats how its designed.
> > Trying to call more than one person at a time might cause more drama than
> > its worth.  Just tell your agents to answer their phones faster... if they
> > dont fire them.
> > 
> > bkw
> > 
> > On Thu, 11 Dec 2003, Markus Mayer wrote:
> > 
> > > Ok, so let me briefly describe our setup and what exactly happens (this
> > > is a test environment):
> > >
> > > 4 phones, 2 logged into the one queue, the other 2 phones used to dial
> > > into the queue.
> > >
> > > One of the non-queue phones dials the queue number, one of the queue
> > > phones rings. Now (while this phone is still ringing) the second
> > > non-queue phone dials the queue also -- but nothing happens. The person
> > > on this phone gets a ring-tone (just like the first), but still only the
> > > first queue-phone actually rings (because of the first call), the second
> > > queue-phone is silent (it should be ringing, though, because of the 2nd
> > > call). So there's 2 people phoning the queue (both hearing ring tones),
> > > 2 phones in the queue ready to answer incoming calls, but only the first
> > > phone is ringing.
> > >
> > > If, however, the first queue-phone (the one that's ringing) is picked
> > > up, then finally the second queue-phone starts ringing, announcing the
> > > 2nd call that's already been waiting in the queue for a while, hearing a
> > > ring tone. Why aren't those two queue phones ringing simultaneously if
> > > there's more than one incoming call (one phone for each call)?
> > >
> > > Note that we want only one phone ringing for each incoming call (so
> > > ringall doesn't work for us), but we also want more than one phone
> > > ringing if there's more than one call coming in at the same time.
> > >
> > > Thanks,
> > > Markus
> > >
> > > On Thu, 2003-12-11 at 14:36, Brancaleoni Matteo wrote:
> > > > so, from queues.conf.sample
> > > >
> > > > A strategy may be specified.  Valid strategies include:
> > > >
> > > > ringall - ring all available channels until one answers (default)
> > > > roundrobin - take turns ringing each available interface
> > > > leastrecent - ring interface which was least recently called by this
> > > > queue
> > > > fewestcalls - ring the one with fewest completed calls from this queue
> > > > random - ring random interface
> > > >
> > > > matteo
> > > >
> > > >
> > > > Il gio, 2003-12-11 alle 23:34, Derek Barber ha scritto:
> > > > > That does get both phones ringing, however that is not the solution
> > > > > problem we are having.  We need the queue to work in leastrecent mode.
> > > > > If there is only one call in the queue only one agent's phone should
> > > > > ring.  However, if there is two calls in the queue then two agent's
> > > > > phones should ring.  Three calls, three phones, etc.  The problem we are
> > > > > having is that if there is two calls in the queue, instead of ringing
> > > > > two agent phones, it only rings one agent's phone.  And thus callers are
> > > > > sitting in the queue when they don't have to be, because there is an
> > > > > available agent to take their call, but that agent's phone doesn't ring.
> > > > >
> > > > > does that explain things better?
> > > > >
> > > > > derek
> > > > >
> > > > > On Thu, 2003-12-11 at 14:14, Brancaleoni Matteo wrote:
> > > > > > strategy=ringall
> > > > > > in queues.conf
> > > > > >
> > > > > > matteo
> > > > > >
> > > > > > Il gio, 2003-12-11 alle 23:18, Derek Barber ha scritto:
> > > > > > > Hello!
> > > > > > >
> > > > > > > We are having an interesting problem with the queue.  What is happening
> > > > > > > is that no matter how many agents are logged into the queue, only one
> > > > > > > phone will ring at one time.  So, for example, if we have two agents in
> > > > > > > the queue and two incoming calls.  The first incoming call will ring on
> > > > > > > one of the agent phones, but the second agent's phone will not ring
> > > > > > > until the first agent picks up the phone.  Both incoming calls are in
> > > > > > > the queue, I have tested and confirmed that, and both agents are in the
> > > > > > > queue and available.  However, only one phone is able to ring at one
> > > > > > > time.
> > > > > > >
> > > > > > > Has anyone seen this problem or know of any solutions?
> > > > > > >
> > > > > > > Thanks,
> > > > > > > Derek
> > >
> > >
> > > 

Re: [Asterisk-Users] MSN messenger and *

[Tilghman Lesher]

On Sunday 21 December 2003 18:12, Craig Waddington wrote:
> I have read the guides on using Messenger to connect via SIP.
>
> I just cant get it to connect, even inside the LAN.
>
> I enter :5036, it trys to sign in, but times out
> and says Service Unavailable.

Port 5036 is for IAX.  You want to use port 5060 for SIP.

-Tilghman

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Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download

[Carlos Arnt]

I put like the readme.txt say the code in my web page , put the OCX in the same Directory, but not work.
Did has any problem ??
 
Thanks
 
The code:
 



New Page 1



	CODEBASE="">



On Sun, 21 Dec 2003 23:46:21 +0200, Dan wrote:> Hi all,>>> A first basic version of DIAX as an ActiveX can be downloaded from:> http://www.laser.com/dante/diax/activediax.zip>> There is only one small file (diax.ocx) and a readme.txt with the> usage instructions.> For the moment you can only place authenticated (or not) calls and> there is no feedback (ring, messages, etc)> Put this simple thing on your web page and you will be able to dial> from any browser.> It works only with IAX2, so you can test it with IAXTEL too> (supported, just needed to dial the number).>> Please send me your feedback and features request for further> development of both standalone and ActiveX versions of DIAX.>> Best regards,> Dan> P.S. Unfortunatelly it works only on IBM PC compatible computers> (not Mac's or Pocket PCs)>> ___> Asterisk-Users mailing list> [EMAIL PROTECTED]> http://lists.digium.com/mailman/listinfo/asterisk-users


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RE: [Asterisk-Users] SNOM 200 and * issues

[Rich Adamson]

> Since this thread may get the attention of snom 200 users I'll also ask
> whether anybody has multiple "lines" registered to Asterisk on firmware
> versions 2.02t or 2.02z ? I tried to config a snom 200 the other day
> with each of the 5 "line" buttons assigned to a line and each line
> registered to Asterisk.
> 
> What I found was that the registrations would fail most of the time and
> the snom 200 would become very slow while the Asterisk console showed
> continuous registration attempts. Deleting the registrations for Lines
> 2-5 allowed line 1 to register.
> 
> On 2.02t I previously noticed the snom 200 often returning BUSY HERE
> when Asterisk tried to place a call to it - the phone was doing nothing.

Yes, I see the same thing by simply defining two lines. I was registering
x3002 on button 1 and x3008 on button 2; sometimes it works, and most of
the time it doesn't.  When it does work, regardless of which extn was
called, the first button would also flash. Therefore, no way for the
phone user to know which extn was actually called. (I'm on v2.02t)

Also ran into a problem where after defining the two extensions, both
button 1 and 2 would lite up and stay on regardless of what the user
would do.

While attempting to diagnose the issue, I also noticed that every other
reboot acted near normal and the odd numbered reboots were useless.
Reboot, test, reboot, same test, reboot, same test, etc, always came
up with differing results.

When the phone is slow to react, I've noticed (using a Sniffer) that hundreds
of packets are flowing between the Snom and Asterisk. The packets are a
constant loop of attempted registration and failures, although once in a
great while it will succeed. (Removing all line definitions accept for
button 1 makes the phone stable again.)

I reported a problem via the snom web site about a week ago, and a support 
person accessed my snom 200. The bottom line, according to him, was "that's 
the way the phone is suppose to work" (in terms of "not" being able to 
identify which extn is ringing). He was going to try replicating the
problem and I've not heard from him since.

For now the snom 200 remains a single-extension phone religate to the
test environment; won't go production like that.

Rich


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Re: [Asterisk-Users] MSN messenger and *

[Balaji NJL]

Do a search on this list for details on how to 
configure MSN. Also post more info like what version u r using and send the 
details of ur sip.conf.
 
-B

  - Original Message - 
  From: 
  Craig 
  Waddington 
  To: [EMAIL PROTECTED] 
  
  Sent: Sunday, December 21, 2003 4:12 
  PM
  Subject: [Asterisk-Users] MSN messenger 
  and *
  
  
  I have read the guides on 
  using Messenger to connect via SIP.
   
  I just cant get it to 
  connect, even inside the LAN.
   
  I enter :5036, it trys to sign in, but times out and says Service 
  Unavailable.
   
  Do I need anything extra 
  in my configs for Messenger to work?
   
  Have * admins managed to 
  get this to work?
   
  Any help 
  welcome.
   
  Thanks 
  
   
   
   
   
   
   

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Re: [Asterisk-Users] MSN messenger and *

[Balaji NJL]

Hi,
 
i am interested. send me more details on ur project 
and where r u located.
 
-Balaji

  - Original Message - 
  From: 
  Craig 
  Waddington 
  To: [EMAIL PROTECTED] 
  
  Sent: Sunday, December 21, 2003 4:12 
  PM
  Subject: [Asterisk-Users] MSN messenger 
  and *
  
  
  I have read the guides on 
  using Messenger to connect via SIP.
   
  I just cant get it to 
  connect, even inside the LAN.
   
  I enter :5036, it trys to sign in, but times out and says Service 
  Unavailable.
   
  Do I need anything extra 
  in my configs for Messenger to work?
   
  Have * admins managed to 
  get this to work?
   
  Any help 
  welcome.
   
  Thanks 
  
   
   
   
   
   
   

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RE: [Asterisk-Users] large implementation

[Todd Wallace]

I am also interested in large scale deployments.

Todd

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Hauser
Sent: Sunday, December 21, 2003 5:45 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] large implementation

Hello,

I am looking at Asterisk for a large scale Implementation, after much
customization of course. But it looks like a nice base. I was wondering
if people could contact me either on the list or off list with examples
of large implementations. In the range of 5K-30K active users.

Best Regards,
David 


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RE: [Asterisk-Users] Extensions Flash Command

[Kevin]

I was able to get it to send DTMF by removing the wait command.  Does
anyone know how I would retrieve a call that is parked and transfer with
the Flash command?  Example call comes in and is transferred to Call
Park.  Have the system call an extension, upon entering a verification
code, the call is removed from Call Park and transferred back on the
originating PBX using the Flash and Ssend DTMF commands.  Any
suggestions on how to retrieve a parked call with the extensions logic?

-Original Message-
From: Kevin [mailto:[EMAIL PROTECTED] 
Sent: Sunday, December 21, 2003 6:42 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Extensions Flash Command

I am trying to use an application that will flash on a ZAP and transfer
a call on the FXO line.  When I use example "1", the command SendDTMF is
never invoked and the call is disconnected.  When I remove the Flash
comman(as in example '2') the SendDTMF works.  Any suggestions as to
what if anything I'm missing here?

Thanks

Example '1'


exten => 2701,1,Playback,transfer
exten => 2701,2,Flash
exten => 2701,3,Wait(1)
exten => 2701,4,SendDTMF(12125551212#)
exten => 2701,5,Hangup

Example '2'

exten => 2701,1,Playback,transfer
;exten => 2701,2,Flash
exten => 2701,2,Wait(1)
exten => 2701,3,SendDTMF(12125551212#)
exten => 2701,4,Hangup


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[Asterisk-Users] MSN messenger and *

[Craig Waddington]

I have read the guides on using Messenger to connect
via SIP.

 

I just cant get it to connect, even inside the LAN.

 

I enter :5036, it trys to
sign in, but times out and says Service Unavailable.

 

Do I need anything extra in my configs for Messenger
to work?

 

Have * admins managed to get this to work?

 

Any help welcome.

 

Thanks 

 

 

 

 

 

 

Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download

[González]

Hi Dan:

I installed your Activex version of DIAX in Delphi 5,
but when i try to use it in a project, i got an error
"OLE error 800A01C5". I'm using windows 98SE.
I hope you can help me.

Thanks in advance.

--- Dan <[EMAIL PROTECTED]> wrote:
> Hi all,
> 
> A first basic version of DIAX as an ActiveX can be
> downloaded from:
> http://www.laser.com/dante/diax/activediax.zip
> 
> There is only one small file (diax.ocx) and a
> readme.txt with the usage
> instructions.
> For the moment you can only place authenticated (or
> not) calls and there is
> no feedback (ring, messages, etc)
> Put this simple thing on your web page and you will
> be able to dial from any
> browser.
> It works only with IAX2, so you can test it with
> IAXTEL too (supported, just
> needed to dial the number).
> 
> Please send me your feedback and features request
> for further development of
> both standalone and ActiveX versions of DIAX.
> 
> Best regards,
> Dan
> P.S. Unfortunatelly it works only on IBM PC
> compatible computers (not Mac's
> or Pocket PCs)
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
>
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Re: [Asterisk-Users] s, h, t, etc, extensions?

[Brian West]

h isn't used for billing it can be used for other stuff but don't rely on
it.  If you say park a call.. and pick it back up. exten => h won't
execute.  Thus you don't want to rely on it for any type of billing or
processing.  Either way the CDR data is still logged correctly without it.

bkw

On Sun, 21 Dec 2003, Olle E. Johansson wrote:

> Thank you all for the explanations.
>
> I've updated the wiki page:
> http://www.voip-info.org/tiki-index.php?page=Asterisk+standard+extensions
>
> /O
>
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[Asterisk-Users] large implementation

[David Hauser]

Hello,

I am looking at Asterisk for a large scale Implementation, after much
customization of course. But it looks like a nice base. I was wondering
if people could contact me either on the list or off list with examples
of large implementations. In the range of 5K-30K active users.

Best Regards,
David 


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[Asterisk-Users] Extensions Flash Command

[Kevin]

I am trying to use an application that will flash on a ZAP and transfer
a call on the FXO line.  When I use example "1", the command SendDTMF is
never invoked and the call is disconnected.  When I remove the Flash
comman(as in example '2') the SendDTMF works.  Any suggestions as to
what if anything I'm missing here?

Thanks

Example '1'


exten => 2701,1,Playback,transfer
exten => 2701,2,Flash
exten => 2701,3,Wait(1)
exten => 2701,4,SendDTMF(12125551212#)
exten => 2701,5,Hangup

Example '2'

exten => 2701,1,Playback,transfer
;exten => 2701,2,Flash
exten => 2701,2,Wait(1)
exten => 2701,3,SendDTMF(12125551212#)
exten => 2701,4,Hangup


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Re: [Asterisk-Users] Dialing dead SIP peers give misleading (BUSY) voicemail result ...

[Darren Nickerson]

> >>Is this the intended result when trying to dial a disconnected SIP
> >>extension, or have I misconfigured something? Does the dialplan above,
which
> >>was built using analog handsets, need to be more intelligent to deal
with
> >>SIP connectivity/registration status?
> >
> >
> > I've noticed the same thing. I don't believe its the desired behaviour
> > but I don't read C-code well enough to fix it either.
>
> Add this check to your stdexten macro:
> exten => s,3,DBget(sipcheck=SIP/Registry/${sipdevicenameinsip_conf})
>
> Check DBget on the wiki. If it fails, there's nothing registred and you
can forward
> directly to unavailable. If it succeeds, there's a current registration
and you can dial as usual.


Olle,

Having thought about this more, ... would you say your solution is

a) a workaround to an Asterisk bug, or
b) a normal part of SIP peer dialing best practice?

Does a SIP peer actually need to be registered with Asterisk to be able to
receive calls? I thought registration was only necessary when there's funky
network stuff in the way, such as NAT. If registration is REQUIRED, it seems
to me that Asterisk should already be doing the check you proposed I add
before it dials out as part of the 'Dial' application, and should simply
move on to the next step in the dialplan if fails this check. If one dials a
SIP peer that is unreachable for any reason (independent of registration
status), shouldn't the failure mode be 'unavailable' instead of 'busy'? Busy
just seems so misleading.

I tried an analogous experiment with a Zap extension ... unplugged it from
the Asterix box and dialed that extension. Asterix rang that extension until
timeout and then rolled over to the unavailable message. That's exactly the
behaviour I would have expected on a SIP dialout to an unreachable peer ...
Dial() would just think it's ringing the remote, but the remote never sees
the traffic cuz it's not in a connected state.

In the case of a network-disconnected SIP peer, the Dial application fails,
routing you to n+101. That's what happens when you dial a busy Zap extension
(I think the same thing also happens when someone's talking on that SIP
peer) making disconnected/unreachable SIP  Dial look like a BUSY. This seems
wrong to my (admittedly still quite naieve) understanding.

In the case of a physically-disconnected ZAP extension, the Dial application
succeeds, moving on to the next step in the dialplan. That is much more in
line with my expectation.

Thoughts?

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 office
+1.215.243.8335 fax

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RE: [Asterisk-Users] modem to modem calls through asterisk

[Don Pobanz]

Replacing my zhone channel bank with an Adtran 750 has fixed my modem - 
modem connect speed problems.

I don't know whether this is something unique to zhone channel banks, 
my particular zhone channel bank or my  installation/configuration. 
Regardless, it is good to have identified the problem.

Don Pobanz

On Tuesday, November 25, 2003 5:52 PM, Don Pobanz 
[SMTP:[EMAIL PROTECTED] wrote:
> Modem connect speeds on calls through * seem to be lower than calls
> made through the telephone company lines or our old Rolm PBX. All 
data
> calls have 2 wire analog modems on both ends.
>
> For my set up I have channels of a Zhone channel bank tied to 2
> modems.
> The Zhone channel bank interfaces my * server with a T400P card.
> modem  --- Zhone Channel bank - * via T400P card - Zhone channel bank
> -
> modem
>
> In extensions.conf I have added the 'd'ata option.
> exten => 333,1, Dial(Zap/19,,d)
>
> In zapata.conf echo cancel has been turned off and I have 
experimented
> with different size jitterbuffers, none of which seem to help.
>
> I couldn't find anything else in the archives. Other suggestions?
>
> Don Pobanz

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Re: [Asterisk-Users] Dialing dead SIP peers give misleading (BUSY) voicemail result ...

[Darren Nickerson]

Thanks for the helpful reply! Without using macros, the following works
well:

exten => 8102,1,DBget(sipcheck=SIP/Registry/${someone})
exten => 8102,3,Dial(SIP/someone,20)
exten => 8102,102,Voicemail(u8102)
exten => 8102,103,Hangup
exten => 8102,104,Voicemail(b8102)
exten => 8102,105,Hangup

Thanks also for the poke I needed to begin looking into macros. The sample
macro-stdexten (which you recommended customizing) looks like:

exten => s,1,Dial(${ARG2},20)
exten => s,2,Voicemail(u${ARG1})
exten => s,3,Goto(default,s,1)
exten => s,102,Voicemail(b${ARG1})
exten => s,103,Goto(default,s,1)

I don't see a way to add the DBGet check for a SIP registration in a generic
way ... if this is a Zap extension there won't be a registration but we
should still try to dial, right? Do I need a macro-stdZAPexten macro that
looks like the one above, and a macro-stdSIPexten that looks something like
(assumes my SIP naming scheme is based on extension number):

;
; ${ARG1} - Name of SIP peer to dial
;
exten => s,1,DBget(sipcheck=SIP/Registry/${ARG1})
exten => s,3,Dial(SIP/${ARG1},20)
exten => s,4,Voicemail(u${MACRO_EXTEN})
exten => s,5,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup

in addition to the macro-stdexten one?

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 office
+1.215.243.8335 fax

- Original Message - 
From:
To: <[EMAIL PROTECTED]>
Sent: Sunday, December 21, 2003 4:48 PM
Subject: Re: [Asterisk-Users] Dialing dead SIP peers give misleading (BUSY)
voicemail result ...


> Rich Adamson wrote:
> >>We have several people using SIP softphones in the office. When they
leave
> >>for the day, they power down their workstations, causing their
registration
> >>with Asterix to quickly timeout.
> >>
> >>Here's the entry for one such extension in extensions.conf:
> >>
> >>exten => 8102,1,Dial(SIP/someone,20)
> >>exten => 8102,2,Voicemail(u8102)
> >>exten => 8102,3,Hangup
> >>exten => 8102,102,Voicemail(b8102)
> >>exten => 8102,103,Hangup
> >>
> >>The desired behaviour when they're gone for the day is to have their
> >>voicemail play the 'unavailable' greeting, and record a message.
> >>Unfortunately, it seems to play the 'busy' greeting instead:
> >>
> >>-- Executing Dial("SIP/darren-0eee", "SIP/someone|20") in new stack
> >>  == Everyone is busy at this time
> >>-- Executing VoiceMail("SIP/darren-0eee", "b8102") in new stack
> >>-- Playing 'voicemail/default/8102/busy' (language 'en')
> >>
> >>Obviously this is misleading to the customer ... the person at that
> >>extension isn't even in the office, let alone busy.
> >>
> >>Is this the intended result when trying to dial a disconnected SIP
> >>extension, or have I misconfigured something? Does the dialplan above,
which
> >>was built using analog handsets, need to be more intelligent to deal
with
> >>SIP connectivity/registration status?
> >
> >
> > I've noticed the same thing. I don't believe its the desired behaviour
> > but I don't read C-code well enough to fix it either.
>
> Add this check to your stdexten macro:
> exten => s,3,DBget(sipcheck=SIP/Registry/${sipdevicenameinsip_conf})
>
> Check DBget on the wiki. If it fails, there's nothing registred and you
can forward
> directly to unavailable. If it succeeds, there's a current registration
and you can dial as usual.
>
> I hope this helps.
> /O
>
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Re: [Asterisk-Users] FWD / Timed out

[Michael Graves]

I get this a lot. It eventually succeeds in registering, but can take a minute or two.

Michael

-Original Message-
From: Isamar Maia <[EMAIL PROTECTED]>
Sent: Dec 21, 2003 10:45 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] FWD / Timed out


Hi Folks,

I can ping fwd.pulver.com with no problem but not
getting to register with them...

NOTICE[4101]: File chan_sip.c, Line 2476 (sip_reg_timeout): Registration
for '[EMAIL PROTECTED]' timed out, trying again


Anybody knows how to fix that?

Thanks,

Isamar


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Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download

[Dan]

Hi,

>- Original Message - 
>From: "Peer Oliver schmidt" <[EMAIL PROTECTED]>
>Subject: Re: [Asterisk-Users] First version of the ActiveX version of DIAX
(0.1.0) available for download
..

> Dan,
>>...
> > P.S. Unfortunatelly it works only on IBM PC compatible computers (not
Mac's
> > or Pocket PCs)
>
> is there a technical reason for not supplying a version running under
> the other OSs, or do you plan on releasing one for Pocket PC as well?
> There are a lot of small Pocket PC available with WLAN and/or Bluetooth,
> which would be ideal as a (D)IAX phone.
I intend to build a PocketPC version too.
It was just a warning that the ActiveX version does run only on Windows
Computers.
Even if you have a browser on a Mac or other type of computer, the low level
library (integrated in the ActiveX)  is not compatible, so it does not work.

Best regards,
Dan


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Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download

[Peer Oliver schmidt]

Dan,
A first basic version of DIAX as an ActiveX can be downloaded from:
http://www.laser.com/dante/diax/activediax.zip
[..]
P.S. Unfortunatelly it works only on IBM PC compatible computers (not Mac's
or Pocket PCs)
is there a technical reason for not supplying a version running under 
the other OSs, or do you plan on releasing one for Pocket PC as well? 
There are a lot of small Pocket PC available with WLAN and/or Bluetooth, 
which would be ideal as a (D)IAX phone.

rgds
pos
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RE: [Asterisk-Users] SNOM 200 and * issues

[Richard Alexander]

Since this thread may get the attention of snom 200 users I'll also ask
whether anybody has multiple "lines" registered to Asterisk on firmware
versions 2.02t or 2.02z ? I tried to config a snom 200 the other day
with each of the 5 "line" buttons assigned to a line and each line
registered to Asterisk.

What I found was that the registrations would fail most of the time and
the snom 200 would become very slow while the Asterisk console showed
continuous registration attempts. Deleting the registrations for Lines
2-5 allowed line 1 to register.

On 2.02t I previously noticed the snom 200 often returning BUSY HERE
when Asterisk tried to place a call to it - the phone was doing nothing.


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Re: [Asterisk-Users] Dialing dead SIP peers give misleading (BUSY) voicemail result ...

[Olle E. Johansson]

Rich Adamson wrote:
We have several people using SIP softphones in the office. When they leave
for the day, they power down their workstations, causing their registration
with Asterix to quickly timeout.
Here's the entry for one such extension in extensions.conf:

exten => 8102,1,Dial(SIP/someone,20)
exten => 8102,2,Voicemail(u8102)
exten => 8102,3,Hangup
exten => 8102,102,Voicemail(b8102)
exten => 8102,103,Hangup
The desired behaviour when they're gone for the day is to have their
voicemail play the 'unavailable' greeting, and record a message.
Unfortunately, it seems to play the 'busy' greeting instead:
   -- Executing Dial("SIP/darren-0eee", "SIP/someone|20") in new stack
 == Everyone is busy at this time
   -- Executing VoiceMail("SIP/darren-0eee", "b8102") in new stack
   -- Playing 'voicemail/default/8102/busy' (language 'en')
Obviously this is misleading to the customer ... the person at that
extension isn't even in the office, let alone busy.
Is this the intended result when trying to dial a disconnected SIP
extension, or have I misconfigured something? Does the dialplan above, which
was built using analog handsets, need to be more intelligent to deal with
SIP connectivity/registration status?
 
I've noticed the same thing. I don't believe its the desired behaviour
but I don't read C-code well enough to fix it either.
Add this check to your stdexten macro:
exten => s,3,DBget(sipcheck=SIP/Registry/${sipdevicenameinsip_conf})
Check DBget on the wiki. If it fails, there's nothing registred and you can forward
directly to unavailable. If it succeeds, there's a current registration and you can 
dial as usual.
I hope this helps.
/O
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[Asterisk-Users] FWD / Timed out

[Isamar Maia]

Hi Folks,

I can ping fwd.pulver.com with no problem but not
getting to register with them...

NOTICE[4101]: File chan_sip.c, Line 2476 (sip_reg_timeout): Registration
for '[EMAIL PROTECTED]' timed out, trying again


Anybody knows how to fix that?

Thanks,

Isamar


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[Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download

[Dan]

Hi all,

A first basic version of DIAX as an ActiveX can be downloaded from:
http://www.laser.com/dante/diax/activediax.zip

There is only one small file (diax.ocx) and a readme.txt with the usage
instructions.
For the moment you can only place authenticated (or not) calls and there is
no feedback (ring, messages, etc)
Put this simple thing on your web page and you will be able to dial from any
browser.
It works only with IAX2, so you can test it with IAXTEL too (supported, just
needed to dial the number).

Please send me your feedback and features request for further development of
both standalone and ActiveX versions of DIAX.

Best regards,
Dan
P.S. Unfortunatelly it works only on IBM PC compatible computers (not Mac's
or Pocket PCs)

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Re: [Asterisk-Users] Callwaiting / limits?

[Paul Liew]

- Original Message - 
From: "Stephen J. Wilcox" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, December 22, 2003 12:54 AM
Subject: [Asterisk-Users] Callwaiting / limits?


> Hi,
>  I'm using grandstream phones, when on a call and a second call comes in
the
> call waiting indication is to play ringing which means you cant actually
hear
> your original call. I want to stop this but cant, heres my options
>
> 1. Change the callwaiting indication, I assume this is produced by the
phone and
> in the case of grandstream there seems to be no way to control this.
>
> 2. Use of incoming/outgoing limit in sip.conf. This works okay except
there is
> no 'absolute limit' type option, meaning that if i place an outbound call
from
> my grandstream it is possible to send a new incoming call in and we have
the
> call waiting again.
>
> I assume others have found this, whats the solution?
>
> Steve
>

Hi Steve,

The incominglimit applies to both incoming and outgoing calls, so long as
I'm on the phone, any incoming call gets sent to voicemail. Use the "sip
show inuse" on the CLI to check the inuse counter is being incremented when
on a call, whether receiving or outgoing.

Is anybody else having this problem ?

Paul

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Re: [Asterisk-Users] Dialing dead SIP peers give misleading (BUSY) voicemail result ...

[Rich Adamson]

> We have several people using SIP softphones in the office. When they leave
> for the day, they power down their workstations, causing their registration
> with Asterix to quickly timeout.
> 
> Here's the entry for one such extension in extensions.conf:
> 
> exten => 8102,1,Dial(SIP/someone,20)
> exten => 8102,2,Voicemail(u8102)
> exten => 8102,3,Hangup
> exten => 8102,102,Voicemail(b8102)
> exten => 8102,103,Hangup
> 
> The desired behaviour when they're gone for the day is to have their
> voicemail play the 'unavailable' greeting, and record a message.
> Unfortunately, it seems to play the 'busy' greeting instead:
> 
> -- Executing Dial("SIP/darren-0eee", "SIP/someone|20") in new stack
>   == Everyone is busy at this time
> -- Executing VoiceMail("SIP/darren-0eee", "b8102") in new stack
> -- Playing 'voicemail/default/8102/busy' (language 'en')
> 
> Obviously this is misleading to the customer ... the person at that
> extension isn't even in the office, let alone busy.
> 
> Is this the intended result when trying to dial a disconnected SIP
> extension, or have I misconfigured something? Does the dialplan above, which
> was built using analog handsets, need to be more intelligent to deal with
> SIP connectivity/registration status?
 
I've noticed the same thing. I don't believe its the desired behaviour
but I don't read C-code well enough to fix it either.

Rich


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[Asterisk-Users] 486 Busy message - SNOM 200

[tony banks]

Hello All,

Whenever I try calling SNOM 200, I am getting "Everyone is Busy at this time" message 
though line is free. I tried with firmware image of
2.02t and 2.03b (this I received from SNOM customer support) but problem still exists. 
Rebooting Phone doesn't help either.

So, is it a firmware problem or there might be problem with the my asterisk 
configuration.

Thanks for your time.

Tony

[Asterisk-Users] Dialing dead SIP peers give misleading (BUSY) voicemail result ...

[Darren Nickerson]

Folks,

We have several people using SIP softphones in the office. When they leave
for the day, they power down their workstations, causing their registration
with Asterix to quickly timeout.

Here's the entry for one such extension in extensions.conf:

exten => 8102,1,Dial(SIP/someone,20)
exten => 8102,2,Voicemail(u8102)
exten => 8102,3,Hangup
exten => 8102,102,Voicemail(b8102)
exten => 8102,103,Hangup

The desired behaviour when they're gone for the day is to have their
voicemail play the 'unavailable' greeting, and record a message.
Unfortunately, it seems to play the 'busy' greeting instead:

-- Executing Dial("SIP/darren-0eee", "SIP/someone|20") in new stack
  == Everyone is busy at this time
-- Executing VoiceMail("SIP/darren-0eee", "b8102") in new stack
-- Playing 'voicemail/default/8102/busy' (language 'en')

Obviously this is misleading to the customer ... the person at that
extension isn't even in the office, let alone busy.

Is this the intended result when trying to dial a disconnected SIP
extension, or have I misconfigured something? Does the dialplan above, which
was built using analog handsets, need to be more intelligent to deal with
SIP connectivity/registration status?

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 office
+1.215.243.8335 fax

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Re: [Asterisk-Users] IVR Configuration in *

[Andrew Kohlsmith]

> Asterisk IVR gets a lot of results from the Asterisk mailing list on
> search.voip-forum.com
 
> ...a good place to start searching for VoIP and Asterisk

Well yes I bet it would be...  unfortunately I don't have "voip:" defined in 
my Konqueror shortcuts.  "gg:asterisk ivr example" was the fast way for 
me.  :-)

Regards,
Andrew
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[Asterisk-Users] Nortel IP phones?

[C. Johnson]

Howdy-

Has anyone on the list implemented Nortel IP
telephones (the exact name
escapes me at the moment) with *?

Doing some work for a small trading firm which
needs a phone to have
the ability to dial into a improvised
hoot-n-holler system (* conference),
so it needs speakerphone, and multi-line
capability.

Also, I'm open to cisco phones, going to look at
those now.

Thx,
-cedrick


===
Cedrick Johnson
www.cedrick.net

Market Commentary:
http://uranium235.blogspot.com
Charts: http://charts.cedrick.net
"You could starve at a banquet if you
were afraid the food was poisioned"
- Art Cashin, CNBC interview

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Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[Darren Nickerson]

Thanks Kevin!

My overall defaults in sip.conf are:

allow=ulaw  ; Allow codecs in order of preference
allow=ilbc
allow=gsm
allow=a_mu

So I would have thought that ulaw codec would have been chosen if available.
Just in case, I added the following to my SJphone test entry in sip.conf:

[darren]
type=friend
host=dynamic
nat=yes
dtmfmode=inband
username=1234
secret=1234
disallow=all
allow=ulaw

I restarted Asrterisk, and re-registered SJPhone with it ... but I still see
the same problem.

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 office
+1.215.243.8335 fax

- Original Message - 
From: "Kevin Bockman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, December 21, 2003 2:05 PM
Subject: Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...


> >I'll try to clarify. Over a series of several logins to voicemail
>entering 1234 for username and password, here's what I see:
> >
> >-- Incorrect password '1f123f344' for user '11223344' (context =
>)
> >-- Incorrect password '11223f344' for user '11223f344' (context =
>)
> >-- Incorrect password '112f23f344' for user '1122334f4' (context =
> >)
> >-- Incorrect password '1f123344' for user '1f12334f4' (context =
>)
> >-- Incorrect password '123f344' for user '12334f4' (context = any>)
>
> I've never seen an f included, or had a double digit problem with SJPhone.
Here's what I'm using.
>
> In general:
>
> disallow=all
> allow=ulaw
>
> [test]
> type=friend
> secret=test
> context=default
> host=dynamic
> callerid="blah" <1234>
> ;mailbox=1234
> dtmfmode=inband
>
> That's the only dtmfmode that SJPhone supports.
>
> Kevin
>
> _
> Are you a Techie? Get Your Free Tech Email Address Now! Visit
http://www.TechEmail.com
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>

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Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[Kevin Bockman]

>I'll try to clarify. Over a series of several logins to voicemail >entering 1234 for 
>username and password, here's what I see:
>
>-- Incorrect password '1f123f344' for user '11223344' (context = >)
>-- Incorrect password '11223f344' for user '11223f344' (context = >)
>-- Incorrect password '112f23f344' for user '1122334f4' (context =
>)
>-- Incorrect password '1f123344' for user '1f12334f4' (context = >)
>-- Incorrect password '123f344' for user '12334f4' (context = any>)

I've never seen an f included, or had a double digit problem with SJPhone.  Here's 
what I'm using.

In general:

disallow=all
allow=ulaw

[test]
type=friend   
secret=test
context=default
host=dynamic
callerid="blah" <1234>
;mailbox=1234
dtmfmode=inband

That's the only dtmfmode that SJPhone supports.

Kevin

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Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[Darren Nickerson]

John,

Having started this thread, I guess I should comment.

While a bug may exist that (simply and only) doubles up DTMF digits (as
others have reported in the case of Grandstream? phones), I cannot reproduce
that exact behaviour with the soft-phone SJphone product I'm using.

I'll try to clarify. Over a series of several logins to voicemail entering
1234 for username and password, here's what I see:

-- Incorrect password '1f123f344' for user '11223344' (context = )
-- Incorrect password '11223f344' for user '11223f344' (context = )
-- Incorrect password '112f23f344' for user '1122334f4' (context =
)
-- Incorrect password '1f123344' for user '1f12334f4' (context = )
-- Incorrect password '123f344' for user '12334f4' (context = )

As you can see, the digits are commonly doubled, but not always. And what's
up with that f??

I'm happy to (and motivated to) look into this more deeply, but I'm
relatively new to Asterisk and not quite certain how to go about
troubleshooting/debugging this. I certainly don't feel I know enough now to
point the finger at Asterisk and open a bug - I'm still thinking it's
possible I've goofed up some config somewhere along the line.

Does anyone have DTMF detection working over SIP with a softphone product
running on Windows?

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 office
+1.215.243.8335 fax


- Original Message - 
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, December 21, 2003 10:22 AM
Subject: Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...


> So, it seems a new bug has been found, which may or may not be at the
> root of this problem.
>
> Let me describe it, and see if you agree with the synopsis:
>
>Asterisk, despite having dtmfmode= set to a particular value in
> sip.conf for a peer, will listen for SIP Info method transmissions
> even if RFC2833 is selected.  In some phones (Grandstream, in
> particular) this causes double-transmission of digits, since the
> phone sends both types of DTMF transmissions without blocking the
> other.  Asterisk should ignore the other two types of DTMF
> transmission when selected to do one type of reception to counter
> these types of equiment peculiarities which seem to prevent correct
> DTMF usage.
>
>
> If I have described this correctly (I don't know - I don't have
> visibility into this problem) then can someone else (preferably
> someone with the problem) open a ticket?
>
> JT
>
>
> >I had the same problem with Grandsteam phones and *.  No other hard
> >or soft phones have the 'double digit' problem with *.  I don't
> >think Asterisk can do both RFC2833 and in-band DTMF at the same
> >time.  It does, however, do RFC2833 and SIP Info at the same time
> >(SIP Info method seems to be on all the time, even when RFC2833 is
> >selected in the sip.conf file).  Switching the Grandsteam to SIP
> >Info allowed it to talk to Asterisk and fixed the double digits
> >problem.
> >
> >- Jim
> >
> >Chris Albertson wrote:
> >
> >>I think this is a problem on the Asterisk side.  I'm seeing
> >>the same problem using a Grandstream Budgetone 100.  And the GS
> >>does have setting for both in-band and RFC2833.
> >>
> >>My guess is asterisk is accepting the DTMF tone __both__ ways
> >>It is reading the RFC28833 stuff _and_ "hearing" the audio tones
> >>as well.
> >>
> >>--- Tilghman Lesher
>
>><[EMAIL PROTECTED]
om>
> >>wrote:
> >>
> >>>On Sunday 21 December 2003 00:29, Darren Nickerson wrote:
> >>>
> Folks,
> 
> I can't seem to get DTMF signaling working properly using SJphone
> connecting to Asterisk via a SIP connection. Here's an example of a
> voicemail session where I entered 1234 for both the username and
> the password:
> 
>  -- Incorrect password '11223344' for user '11223f344' (context
> 
> 
> >>
> >>
> >>>Changing the DTMF mode would indeed seem to be the logical
> >>>solution.  However, it appears that SJphone does not support that
> >>>option (after a quick perusal of their PDF).  You might want to file
> >>>a bugtracker request on their website to implement that functionality.
> >>>
> >>
> >>=
> >>Chris Albertson
> >>   Home:   310-376-1029
> >>[EMAIL PROTECTED]
> >>   Cell:   310-990-7550
> >>   Office: 310-336-5189
>
>>[EMAIL PROTECTED]
> >   KG6OMK
> >
> >--
>
>+--
-+
> >| Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC
|
>
>+--
-+
> >| "I never let my schooling get in the way of my education." - Mark Twain
|
> >| "UNIX was never designed to keep people from doing stupid things,
because |
> >|  that policy would also keep them from doing clever things." - Doug
Gwyn  |
> >| "Cool is only three letters

Re: [Asterisk-Users] ivr key press?

[Rich Adamson]

Steve,

Thank you very much for taking the time to remove the forest. Until
one attempts something like this, one does not have an appreciation
for the real impact of contexts and how they interact.

My initial dialplan was a very simple single-context version that started
from John Todd's basic get-started samples (which were fine), but now
the level of sophistication has grown far beyond that.

I think I now might have enough of an understanding to rewrite the entire
extensions.conf file to something much better. Thanks again.

Rich


> On Sun, 2003-12-21 at 10:19, Rich Adamson wrote:
> > > > for the full 3000? (That's that part I'm not seeing; forest and the trees 
> > > > kind of thing).
> > > 
> > > Okay, please step out of the forest.
> > 
> > Okay, some of the forest is beginning to disappear and I can see some trees. ;)
> > I'm getting closer at least. Here's the ivr context as it stands right now,
> > knowing full well that its incomplete. I've having a brain-fart on part of
> > this. In the initial inbound ivr context [inbound-bus2], I press an 8 to
> > goto the [npilist] context where a message is played that says "for Rich,
> > dial extension 3000, etc". When the user enters 3000, what statement needs to
> > follow the exten => s,2,Background(npi-directory) to send the call to the
> > entered extension?  (Note: all of my extensions are defined in elsewhere
> > in [from-sip].)
> 
> In each of your contexts where you wish someone to be able to dial an
> extension directly, you will need to add a include => from-sip. Be
> careful here as you may have included outside dialing functionality in
> that context guessing from the name of the context. 
> 
> > Also, is there a "set verbose" mode or something like that, that would
> > provide some CLI feedback as to return values, etc? (Example, with the
> > menues above, during the message played in [npilist] I'm expecting to
> > enter 3000. However, as soon as I press the "3" I get a "-- Hungup 'Zap/2-1'"
> > which doesn't help me understand what is really going on. Is there
> > another debug mode that would help?
> 
> If you run asterisk with multiple v options you should see a bit. My
> guess is if you had a i extension you might have noticed this earlier. i
> is for invalid, and I believe you can still string together the ${exten}
> and some other stuff to inform the caller what they had pressed that was
> invalid.
> 
> Just to help speed things up and help come out of the forest, here is a
> skeleton that should help you get started.
> Well it started as a skeleton, but I just kept filling in details I
> didn't want to skip on and I nearly filled it all in for you. Hopefully
> this will help.
> 
> [inbound-bus2]
> include => bus2-extensions
> exten => s,1,wait(1)
> exten => s,2,answer
> exten => s,3,digittimeout(5)
> exten => s,4,responsetimeout(10)
> exten => s,5,Background(npi-greeting)
> 
> exten => 1,1,goto(sales,s,1)
> exten => 2,1,goto(custserv,s,1)
> exten => 8,1,goto(npilist,s,1)
> exten => 9,1,voicemailmain2
> 
> exten => i,1,saydigit(${EXTEN})
> exten => i,2,backgroud(invalid)
> 
> [outbound-bus2]
> exten => _9NXX,1,setaccount(npi) ; if needed this is nice
> exten => _9NXX,2,Dial(Zap,g2,${EXTEN:1})
> 
> exten => _9NXXNXX,1,setaccount(npi)
> exten => _9NXXNXX,2,Dial(Zap,g2,${EXTEN:1})
> 
> exten => _99XXNXX,1,playback(No-900-service); needs more specific
>   ;match as there are 9XX area codes.
> 
> [sales]
> include => bus2-extensions
> exten => s,1,goto(bus2-extensions,3000,1)
> 
> [custserv]
> include => bus2-extensions
> exten => s,1,goto(bus2-extensions,3002,1)
> 
> [npilist]
> include => bus2-extensions
> exten => s,1,Wait,1
> exten => s,2,background(npi-directory)
> 
> exten => *,1,goto(inbound-bus2,s,1)
> exten => #,1,goto(inbound-bus2,s,1)
> 
> [from-sip]
> include => bus2-extensions
> include => outbound-bus2
> 
> [bus2-extensions]
> exten => 3001,1,Dial(SIP/3001,15,t)
> exten => 3001,2,voicemail2([EMAIL PROTECTED])
> exten => 3001,3,hangup
> exten => 3001,102,voicemail2([EMAIL PROTECTED])
> exten => 3001,103,hangup
> 
> exten => 3002,1,Dial(SIP/3002,15,t)
> exten => 3002,2,voicemail2([EMAIL PROTECTED])
> exten => 3002,3,hangup
> exten => 3002,102,voicemail2([EMAIL PROTECTED])
> exten => 3002,103,hangup
> 
> -- 
> Steven Critchfield <[EMAIL PROTECTED]>
> 
> ___
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---End of Original Message-


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Re: [Asterisk-Users] IVR Configuration in *

[Olle E. Johansson]

Andrew Kohlsmith wrote:
I won't point you off to google, as "asterisk IVR" and "asterisk IVR 
example" did not turn up anything immediately obvious.
http://search.voip-forum.com/cgi-bin/htsearch?words=asterisk%20ivr&config=voipsearch

Asterisk IVR gets a lot of results from the Asterisk mailing list on search.voip-forum.com

...a good place to start searching for VoIP and Asterisk
/O
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[Asterisk-Users] Strange * & ATA behaviour when directly connected

[Dan]

Hi,

This is my  very strange problem:

I have connected * server and ATA186 directly through a cross ethernet cable
(isolated from the rest of the network).
I have two extensions (101 and 103) connected to the ATA.
When an incoming call on X100, both phones must ring.
This is what I have in the inbound context:

exten => s,1,Dial(SIP/101&SIP/103,60,tTr)
...
When an incoming call arrives, just extension 101 rings and for the other
one I get the following in the * console:
SIP/103-e532 is circuit-busy

I can place internal calls without any problems in both directions
(101-103).

If I change the line in extensions.conf like that:
exten => s,1,Dial(SIP/103&SIP/101,60,tTr)
...
extension 103 rings, but 101 now gives:
SIP/101-e513 is circuit-busy

Why changing the order in the dial string gives me this issue?

Connecting the * and ATA in the local network, this problem disappear.

Anyone knows the cause?

Thanks,
Dan


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RE: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[mikeu]

I have the same "key bounce" problem with a Budgetone 101.  After using the
ZapBarge application to monitor the audio channel I determined that the 101
is pulsing the DTMF tones as long as the key is depressed at a rate of 200mS
or so.  If you tap the key pad quickly only one cycle is transmitted.  I
don't understand this "feature".  Does anyone?  Anyway to disable it?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Sunday, December 21, 2003 9:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

So, it seems a new bug has been found, which may or may not be at the 
root of this problem.

Let me describe it, and see if you agree with the synopsis:

   Asterisk, despite having dtmfmode= set to a particular value in 
sip.conf for a peer, will listen for SIP Info method transmissions 
even if RFC2833 is selected.  In some phones (Grandstream, in 
particular) this causes double-transmission of digits, since the 
phone sends both types of DTMF transmissions without blocking the 
other.  Asterisk should ignore the other two types of DTMF 
transmission when selected to do one type of reception to counter 
these types of equiment peculiarities which seem to prevent correct 
DTMF usage.


If I have described this correctly (I don't know - I don't have 
visibility into this problem) then can someone else (preferably 
someone with the problem) open a ticket?

JT


>I had the same problem with Grandsteam phones and *.  No other hard 
>or soft phones have the 'double digit' problem with *.  I don't 
>think Asterisk can do both RFC2833 and in-band DTMF at the same 
>time.  It does, however, do RFC2833 and SIP Info at the same time 
>(SIP Info method seems to be on all the time, even when RFC2833 is 
>selected in the sip.conf file).  Switching the Grandsteam to SIP 
>Info allowed it to talk to Asterisk and fixed the double digits 
>problem.
>
>- Jim
>
>Chris Albertson wrote:
>
>>I think this is a problem on the Asterisk side.  I'm seeing
>>the same problem using a Grandstream Budgetone 100.  And the GS
>>does have setting for both in-band and RFC2833.
>>
>>My guess is asterisk is accepting the DTMF tone __both__ ways
>>It is reading the RFC28833 stuff _and_ "hearing" the audio tones
>>as well. 
>>
>>--- Tilghman Lesher 
>><[EMAIL PROTECTED]
om> 
>>wrote:
>>
>>>On Sunday 21 December 2003 00:29, Darren Nickerson wrote:
>>>
Folks,

I can't seem to get DTMF signaling working properly using SJphone
connecting to Asterisk via a SIP connection. Here's an example of a
voicemail session where I entered 1234 for both the username and
the password:

 -- Incorrect password '11223344' for user '11223f344' (context
  

>>
>>
>>>Changing the DTMF mode would indeed seem to be the logical
>>>solution.  However, it appears that SJphone does not support that
>>>option (after a quick perusal of their PDF).  You might want to file
>>>a bugtracker request on their website to implement that functionality.
>>>
>>
>>=
>>Chris Albertson
>>   Home:   310-376-1029  
>>[EMAIL PROTECTED]
>>   Cell:   310-990-7550
>>   Office: 310-336-5189  
>>[EMAIL PROTECTED]
>   KG6OMK
>
>--
>+--
-+
>| Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC
|
>+--
-+
>| "I never let my schooling get in the way of my education." - Mark Twain
|
>| "UNIX was never designed to keep people from doing stupid things, because
|
>|  that policy would also keep them from doing clever things." - Doug Gwyn
|
>| "Cool is only three letters away from Fool" - Mike Muir, Suicyco
|
>| "..Government in its best state is but a necessary evil; in its worst
|
>|  state an intolerable one.." - Thomas Paine, "Common Sense" (1776)
|
>+--
-+
>|   Email:  [EMAIL PROTECTED] 
>ICQ UIN:  1695089 |
>+--
-+
>|  Reply problems ?  Turn off the "sign" function in email prog.  Blame MS.
|
>+--
-+
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Re: [Asterisk-Users] SS7 API & Card Solution

[Steven Critchfield]

On Sun, 2003-12-21 at 02:40, Juan J. Sierralta P. wrote:
> On Sun, 2003-12-21 at 04:10, Ray Burkholder wrote:
> > Is this useful as a bootstrap for getting SS7 to Asterisk?
> > 
> > http://www.sangoma.com/api/p-api-ss7.htm
> 
>   You should check http://www.openss7.org the have an stack and works
> with an special version of digium cards, dunno if is the same HW with
> special drivers but it looks much more * friendly.

Oddly enough, it looks like songoma may be similar to Digium in that
they have written and give away the code to help facilitate hardware
sales. If you download the wanpipe software, it has a GPL license in it.
Of course there is where the trouble starts in that it could be included
in the GPL version, but not the Digium proprietary version.

Also from looking at their site, the T1 and e1 cards are still more
expensive that Digium cards. 

Juan may be right that it would be better to work with the openss7
people. Seems they finally have opened up their code for anyone to
download it instead of their core developers.  
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] ToIP (TDD over IP)

[Joel Maslak]

I didn't know if it would work or not, but I figured I'd try slow-speed
half-duplex TDD over GSM & Vonage.

I called a AGI script I have that speaks to TTYs, by calling from Vonage
to one of my Voicepulse lines.  I don't control the Vonage codec, so I
have no idea what it uses, but I am using GSM for the Voicepulse line.
Everything worked fine - echo canceling didn't cause any trouble (I don't
know if it would have if I did full duplex, though), I didn't lose any
characters, etc.  So for people who have a need for this kind of
technology, I can tell you that it will work.

I'm also curious if anyone else is doing this or if anyone else is using
the Asterisk TDD support.

-- 
Joel
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Re: [Asterisk-Users] ivr key press?

[Steven Critchfield]

On Sun, 2003-12-21 at 10:19, Rich Adamson wrote:
> > > for the full 3000? (That's that part I'm not seeing; forest and the trees 
> > > kind of thing).
> > 
> > Okay, please step out of the forest.
> 
> Okay, some of the forest is beginning to disappear and I can see some trees. ;)
> I'm getting closer at least. Here's the ivr context as it stands right now,
> knowing full well that its incomplete. I've having a brain-fart on part of
> this. In the initial inbound ivr context [inbound-bus2], I press an 8 to
> goto the [npilist] context where a message is played that says "for Rich,
> dial extension 3000, etc". When the user enters 3000, what statement needs to
> follow the exten => s,2,Background(npi-directory) to send the call to the
> entered extension?  (Note: all of my extensions are defined in elsewhere
> in [from-sip].)

In each of your contexts where you wish someone to be able to dial an
extension directly, you will need to add a include => from-sip. Be
careful here as you may have included outside dialing functionality in
that context guessing from the name of the context. 

> Also, is there a "set verbose" mode or something like that, that would
> provide some CLI feedback as to return values, etc? (Example, with the
> menues above, during the message played in [npilist] I'm expecting to
> enter 3000. However, as soon as I press the "3" I get a "-- Hungup 'Zap/2-1'"
> which doesn't help me understand what is really going on. Is there
> another debug mode that would help?

If you run asterisk with multiple v options you should see a bit. My
guess is if you had a i extension you might have noticed this earlier. i
is for invalid, and I believe you can still string together the ${exten}
and some other stuff to inform the caller what they had pressed that was
invalid.

Just to help speed things up and help come out of the forest, here is a
skeleton that should help you get started.
Well it started as a skeleton, but I just kept filling in details I
didn't want to skip on and I nearly filled it all in for you. Hopefully
this will help.

[inbound-bus2]
include => bus2-extensions
exten => s,1,wait(1)
exten => s,2,answer
exten => s,3,digittimeout(5)
exten => s,4,responsetimeout(10)
exten => s,5,Background(npi-greeting)

exten => 1,1,goto(sales,s,1)
exten => 2,1,goto(custserv,s,1)
exten => 8,1,goto(npilist,s,1)
exten => 9,1,voicemailmain2

exten => i,1,saydigit(${EXTEN})
exten => i,2,backgroud(invalid)

[outbound-bus2]
exten => _9NXX,1,setaccount(npi) ; if needed this is nice
exten => _9NXX,2,Dial(Zap,g2,${EXTEN:1})

exten => _9NXXNXX,1,setaccount(npi)
exten => _9NXXNXX,2,Dial(Zap,g2,${EXTEN:1})

exten => _99XXNXX,1,playback(No-900-service); needs more specific
;match as there are 9XX area codes.

[sales]
include => bus2-extensions
exten => s,1,goto(bus2-extensions,3000,1)

[custserv]
include => bus2-extensions
exten => s,1,goto(bus2-extensions,3002,1)

[npilist]
include => bus2-extensions
exten => s,1,Wait,1
exten => s,2,background(npi-directory)

exten => *,1,goto(inbound-bus2,s,1)
exten => #,1,goto(inbound-bus2,s,1)

[from-sip]
include => bus2-extensions
include => outbound-bus2

[bus2-extensions]
exten => 3001,1,Dial(SIP/3001,15,t)
exten => 3001,2,voicemail2([EMAIL PROTECTED])
exten => 3001,3,hangup
exten => 3001,102,voicemail2([EMAIL PROTECTED])
exten => 3001,103,hangup

exten => 3002,1,Dial(SIP/3002,15,t)
exten => 3002,2,voicemail2([EMAIL PROTECTED])
exten => 3002,3,hangup
exten => 3002,102,voicemail2([EMAIL PROTECTED])
exten => 3002,103,hangup

-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] FWD via IAXtel?

[Michael Graves]

Hello All,

It seems to me that passing SIP through my NAT router to * is a fair amount of work 
what with patching * and opening router ports, etc. In my setup I only need SIP to 
access FWD numbers. 

I have found that any IAXtel user can dial FWD numbers using the prefix 170099. Brief 
testing shows that this seems to work well for outbound calls to FWD. I may also wich 
to route 800 toll free calls through IAXtel in order to keep my few POTS lines free 
for incomming calls. However, I have heard that IAXtel in not particularily reliable. 
Is this true? Does anyone here make significant use of IAXtel?

Thanks,
Michael




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Re: [Asterisk-Users] ivr key press?

[Rich Adamson]

> > for the full 3000? (That's that part I'm not seeing; forest and the trees 
> > kind of thing).
> 
> Okay, please step out of the forest.

Okay, some of the forest is beginning to disappear and I can see some trees. ;)
I'm getting closer at least. Here's the ivr context as it stands right now,
knowing full well that its incomplete. I've having a brain-fart on part of
this. In the initial inbound ivr context [inbound-bus2], I press an 8 to
goto the [npilist] context where a message is played that says "for Rich,
dial extension 3000, etc". When the user enters 3000, what statement needs to
follow the exten => s,2,Background(npi-directory) to send the call to the
entered extension?  (Note: all of my extensions are defined in elsewhere
in [from-sip].)

[inbound-bus2]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,20
exten => s,5,Background(npi-greeting)  ; "Thanks for calling press 1 for"
   
exten => 1,1,Goto(sales,s,1) 
exten => 2,1,Goto(custserv,s,1)   
exten => 8,1,Goto(npilist,s,1)
exten => 9,1,Goto(vm,s,1)  
  
[sales]  
exten => s,1,Dial(SIP/3001,15,t)  
exten => s,2,Hangup  
   
[custserv]   
exten => s,1,Dial(SIP/3002,15,t)
exten => s,2,Hangup

[npilist]
exten => s,1,Wait,1
exten => s,2,Background(npi-directory)   ; "Press 3000 for Rich..."
exten => s,3,Goto(from-sip,s,1)

[vm]
exten => s,1,VoicemailMain2
exten => s,2,Hangup

Also, is there a "set verbose" mode or something like that, that would
provide some CLI feedback as to return values, etc? (Example, with the
menues above, during the message played in [npilist] I'm expecting to
enter 3000. However, as soon as I press the "3" I get a "-- Hungup 'Zap/2-1'"
which doesn't help me understand what is really going on. Is there
another debug mode that would help?

Rich



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Re: [Asterisk-Users] codec negotiation

[Nguyen Hoang Lan]

Hello Eduardo,

Wednesday, December 17, 2003, 1:08:00 AM, you wrote:

EG> Hi list,

EG> I'm with a little problem on codec negotiation between a cisco827 and
EG> asterisk.

EG> My sip.conf is like that: 

EG> [general]
EG> port = 5060
EG> bindaddr = 0.0.0.0
EG> context = default
EG> amaflags = default
EG> allow=g729
EG> allow=gsm 
EG> allow=alaw
EG> allow=ulaw
EG> ;disallow=all

EG> and cisco like that:

EG> dial-peer voice 6 voip
EG>  destination-pattern 0T
EG>  session protocol sipv2
EG>  session target ipv4:
EG>  dtmf-relay rtp-nte
EG>  codec g711alaw
EG>  no vad   
EG> ! 

EG> When I try to make a call, cisco shows codec g711alaw, but asterisk
EG> shows codec g729A (i have the licenses) and there is no audio. When I
EG> try disallow=g729, the same occurs, but this time asterisk shows codec
EG> gsm.

EG> The only way to make a call is allowing only alaw. But this is not
EG> convenience, since i need to use g279 with another endpoint (working
EG> ok). 

EG> Why this negotiation problem happens?

Try to add to cisco peer (not shown in your mail)

[cisco]

disallow=all
allow=alaw



-- 
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 Nguyenmailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Level(3) SIP termination services

[Alex Rodriguez]

In my experience all dealings with Level 3 require an NDA.

Alex

On Dec 20, 2003, at 11:25 PM, John Todd wrote:

As you speak with providers, so long as there is no NDA, please share 
your experiences with the list.  The medium-sized call-centric shops 
and call centers would benefit greatly from hearing the possible 
competitive alternatives that might be connected to Asterisk systems.
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Re: [Asterisk-Users] DIAX phone busy

[Dan]

Hi,

>- Original Message - 
>From: <[EMAIL PROTECTED]>
>Subject: Re: [Asterisk-Users] DIAX phone busy
>..

> Quite a few of us have noted this problem with not only DIAX but also 
> iaxcomm.  It just seems that no one has figured out what the problem is 
> yet.  Though I believe that there are no shortage of people trying.
> AJ
> 

If someone more experienced in the IAX2 protocol can help,
please try (using the debug version of DIAX) to discover where
the problem is.

Thanks,
Dan
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Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[John Todd]

So, it seems a new bug has been found, which may or may not be at the 
root of this problem.

Let me describe it, and see if you agree with the synopsis:

  Asterisk, despite having dtmfmode= set to a particular value in 
sip.conf for a peer, will listen for SIP Info method transmissions 
even if RFC2833 is selected.  In some phones (Grandstream, in 
particular) this causes double-transmission of digits, since the 
phone sends both types of DTMF transmissions without blocking the 
other.  Asterisk should ignore the other two types of DTMF 
transmission when selected to do one type of reception to counter 
these types of equiment peculiarities which seem to prevent correct 
DTMF usage.

If I have described this correctly (I don't know - I don't have 
visibility into this problem) then can someone else (preferably 
someone with the problem) open a ticket?

JT


I had the same problem with Grandsteam phones and *.  No other hard 
or soft phones have the 'double digit' problem with *.  I don't 
think Asterisk can do both RFC2833 and in-band DTMF at the same 
time.  It does, however, do RFC2833 and SIP Info at the same time 
(SIP Info method seems to be on all the time, even when RFC2833 is 
selected in the sip.conf file).  Switching the Grandsteam to SIP 
Info allowed it to talk to Asterisk and fixed the double digits 
problem.

- Jim

Chris Albertson wrote:

I think this is a problem on the Asterisk side.  I'm seeing
the same problem using a Grandstream Budgetone 100.  And the GS
does have setting for both in-band and RFC2833.
My guess is asterisk is accepting the DTMF tone __both__ ways
It is reading the RFC28833 stuff _and_ "hearing" the audio tones
as well. 

--- Tilghman Lesher 
<[EMAIL PROTECTED]> 
wrote:

On Sunday 21 December 2003 00:29, Darren Nickerson wrote:
   
Folks,

I can't seem to get DTMF signaling working properly using SJphone
connecting to Asterisk via a SIP connection. Here's an example of a
voicemail session where I entered 1234 for both the username and
the password:
-- Incorrect password '11223344' for user '11223f344' (context
 



Changing the DTMF mode would indeed seem to be the logical
solution.  However, it appears that SJphone does not support that
option (after a quick perusal of their PDF).  You might want to file
a bugtracker request on their website to implement that functionality.
=
Chris Albertson
  Home:   310-376-1029  
[EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  
[EMAIL PROTECTED]
  KG6OMK

--
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Re: [Asterisk-Users] DIAX phone busy

[firedude]

Quite a few of us have noted this problem with not only DIAX but also 
iaxcomm.  It just seems that no one has figured out what the problem is 
yet.  Though I believe that there are no shortage of people trying.
AJ


On Sun, 21 Dec 2003 [EMAIL PROTECTED] wrote:

> Yes,I often get the same result, but not always. 
> 
> 
> - Original Message - 
> From: "Michael Welter" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Saturday, December 20, 2003 3:40 PM
> Subject: Re: [Asterisk-Users] DIAX phone busy
> 
> 
> > Yes, I've tried that as well.  When I dial "70" from another extension, 
> > I hear ringing but the DIAX doesn't ring.
> > 
> > Dan wrote:
> > > Hi,
> > > 
> > > 
> > >>Yes, IAX2 is checked.
> > >>
> > > 
> > > Then change the line to:
> > > exten => 70,1,Dial(IAX2/mike,30,tr)
> > > 
> > > 
> > > BR,
> > > Dan
> > > 
> > > 
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > 
> > > 
> > 
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> ___
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> 

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[Asterisk-Users] Callwaiting / limits?

[Stephen J. Wilcox]

Hi,
 I'm using grandstream phones, when on a call and a second call comes in the 
call waiting indication is to play ringing which means you cant actually hear 
your original call. I want to stop this but cant, heres my options

1. Change the callwaiting indication, I assume this is produced by the phone and 
in the case of grandstream there seems to be no way to control this.

2. Use of incoming/outgoing limit in sip.conf. This works okay except there is 
no 'absolute limit' type option, meaning that if i place an outbound call from 
my grandstream it is possible to send a new incoming call in and we have the 
call waiting again.

I assume others have found this, whats the solution?

Steve

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Re: [Asterisk-Users] IVR Configuration in *

[Andrew Kohlsmith]

>  Could you please point me to sources for setting up IVR system using
> asterisk. My goal is that when caller calls, four or five options are
> presented and inputs are accepted from the caller and then appropriate
> actions are taken depending on the inputs from the user.

The Asterisk handbook is a great starting point, and (IIRC) includes 
examples for IVR.  It's located at http://www.asterisk.org/
index.php?menu=support, under the Handbook Project (just under the Google 
search on the page).  

There is also a full-blown IVR example in the default extensions.conf which 
was provided when you installed asterisk.

You might want to read through the support page, as it also includes the 
Wiki, unofficial forums and other sources which would have answered your 
question faster than asking a question like this on the mailing list.  :-)

I won't point you off to google, as "asterisk IVR" and "asterisk IVR 
example" did not turn up anything immediately obvious.

Regards,
Andrew
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RE: [Asterisk-Users] ivr key press?

[David J Carter]

Hi Rich,

For what it's worth her is an example of my IVR.

Hope it helps.

[mainmenu]
;
;"main menu" context with submenu
;
exten => s,1,Answer
include => default ; Main dialplan
;exten => s,2,SayDigits(${CALLERID})
exten => s,3,Background(hello_and_thank_you)
exten => s,4,Wait,t,2
exten => s,5,Goto(options,s,1)
;
[options]
;
exten => s,1,Background(double_glazing)
exten => s,2,SayDigits(444)
exten => s,3,Background(now)
exten => s,4,Background(kitchen_sales)
exten => s,5,SayDigits(555)
exten => s,6,Background(now)
exten => s,7,Background(insurance_sales)
exten => s,8,SayDigits(666)
exten => s,9,Background(now)
;
include => default ; Main dialplan for other extensions
;
exten => 444,1,Goto(glazing,s,1)
exten => 444,2,Hangup
;
exten => 555,1,Goto(kitchen,s,1)
exten => 555,2,Hangup
;
exten => 666,1,Goto(insurance,s,1)
exten => 666,2,Hangup
;
[glazing]
exten => s,1,Background(glazing_thanks)
exten => s,2,MusicOnHold(default)
exten => s,3,Background(sorry_for_delay)
exten => s,4,Goto(s,2)
exten => s,5,Hangup
;
[kitchens]
exten => s,1,Background(kitchen_thanks)
exten => s,2,MusicOnHold(default)
exten => s,3,Background(sorry_for_delay)
exten => s,4,Goto(s,2)
exten => s,5,Hangup
;
[insurance]
exten => s,1,Background(insurance_thanks)
exten => s,2,MusicOnHold(default)
exten => s,3,Background(sorry_for_delay)
exten => s,4,Goto(s,2)
exten => s,5,Hangup
;


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson
Sent: 21 December 2003 03:07
To: Asterisk-a-users-list
Subject: [Asterisk-Users] ivr key press?

I'm testing an ivr implementation (first time) using:
exten => 620,1,Wait,1
exten => 620,2,Answer
exten => 620,3,DigitTimeout,5
exten => 620,4,ResponseTimeout,10
exten => 620,5,Background(npi-greeting)  ; "Thanks for calling press 1 for"

exten => 1,1,Goto(npi-directory,s,1)

For initial testing, I've arbitrarily mapped this onto ext 620 (will
change that later when things are working as expected).

The initial "npi-greeting" message essentially says "...if you know
your party's extension, you can dial it at any time.  Press 1 for
Sales, etc."

If during this initial greeting I press 3000 (which is a valid extn),
I can only press the first "3" before I get kicked out (I can't dial
the full 3000).

Am I supposed to be setting this up to expect only single-digit
key presses (instead of 3000), or am I missing something that would
suggest waiting for four key presses?



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Re: [Asterisk-Users] DIAX phone busy

[Dan]

Hi,

>- Original Message - 
>From: "Michael Welter" <[EMAIL PROTECTED]>
>
> Yes, I've tried that as well.  When I dial "70" from another extension,
> I hear ringing but the DIAX doesn't ring.

Ok. Then it is about the LIBIAX2 bug.
Try using IAX till this will be solved (uncheck IAX2 in the registration
page).

BR,
Dan


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Re: [Asterisk-Users] hangup and billing /*New subject */

[Olle E. Johansson]

John Todd wrote:
At 6:32 PM -0600 12/20/03, Brian West wrote:


On a side note.. you can't use exten => h, if you have any hope of 
getting
accurate billing info.  Its wise to call ResetCDR(w) in your exten => h,
or not use it at all.
> Care to expound a bit on that topic for the wiki, with some details as
> to why?
>
I found out that if you use the 'h' extension, the last application in the cdr is
the 'hangup'. Not very useful. I don't know if this is what Brian is referring to,
so I'm also curios. Brian, plz?
/O

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[Asterisk-Users] IVR Configuration in *

[tony banks]

Hello All,

Could you please point me to sources for setting up IVR system using asterisk. My goal 
is that when caller calls, four or five options are presented and inputs are accepted 
from the caller and then appropriate actions are taken depending on the inputs from 
the user.

Thanks for your time.
Tony

Re: [Asterisk-Users] ivr key press?

[Olle E. Johansson]

As I understand it, building a menu is best done in a separate context, done for the menu.

I'm guessing a bit, but my tests indicate that Asterisk finds out the longest 
extension within
the context (including included contexts) and listens until you've entered as many 
digits as the
longest one, or a DigitTimeout. So if you have a menu with a "1" and one "3000" as the 
longest one,
entering 3000 makes Asterisk stop listening and start processing directly, but entering 
"1" makes
Asterisk wait until a DigitTimeout. User's appreciate fast processing, so try to keep 
your
menues consistent in input length.
The 'i' extension is used for invalid input, catch it. It has to be within the menu 
extension.
The 't' extension is for timeouts. Catch that too.
Rich's example:
> > exten => 620,1,Wait,1
> > exten => 620,2,Answer
> > exten => 620,3,DigitTimeout,5
> > exten => 620,4,ResponseTimeout,10
> > exten => 620,5,Background(npi-greeting)  ; "Thanks for calling press 1 for"
> >
> > exten => 1,1,Goto(npi-directory,s,1)
I would change this to something like

exten => 620,1,Goto(richsmenu,s,1)

[richsmenu]
exten => s,1,wait(1)
exten => s,2,Answer
exten => s,3,DigitTimeout(5)
exten => s,4,ResponseTimeout(10)
exten => s,5,Background(npi-greeting)
;Possible choices
exten => 1,1, blablabla
exten => 2,1, blablabla
;Catch three character extensions
exten => _XXX,1,goto(myextensions,${exten},1
;Invalid extensions
exten => i, 1, Playback(invalid)
exten => i, 2, goto(s,5)
;Timeouts, restart
exten => t, 1, goto (s,5)
Also make sure there's an 'i' extension in [myextensions]

Of course one could include 'myextensions' but there might be other stuff in there 
that you
won't want to be reachable from a menu.
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+BackGround
http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu
Good luck building menues! It's fun.
/Olle
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Re: [Asterisk-Users] SS7 API & Card Solution

[Juan J. Sierralta P.]

On Sun, 2003-12-21 at 04:10, Ray Burkholder wrote:
> Is this useful as a bootstrap for getting SS7 to Asterisk?
> 
> http://www.sangoma.com/api/p-api-ss7.htm

You should check http://www.openss7.org the have an stack and works
with an special version of digium cards, dunno if is the same HW with
special drivers but it looks much more * friendly.

-- 
Juanjo sin .sig

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Re: [Asterisk-Users] s, h, t, etc, extensions?

[Olle E. Johansson]

Thank you all for the explanations.

I've updated the wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+standard+extensions
/O

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Re: [Asterisk-Users] asterisk on beowulf cluster

[Olle E. Johansson]

Steven Critchfield wrote:
On Sat, 2003-12-20 at 14:54, Balaji NJL wrote:

Hi All,

Can i install * on a beowulf cluster or Is * compatible to clusters. I
am planning to install a 4 node beowulf cluster using few cheap
hardwares. If no one had tried before i can spend some time on
installing and configuring * on this cluster. Let me know.


Just a bit of research would let you know that cluster can't migrate
threads that use shared memory. Asterisk is such an app. So no asterisk
wouldn't work on a cluster.
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
FAQ Updated.
/O

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RE: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[Girish Gopinath]

Hi,

I  am using SJPhone here for testing ivr with Asterisk. I haven't seen any 
problem here yet.
I have tried different things for that and finally got it working. I am not 
an expert to explain more about that, but here is the general section form 
my sip.conf. dont know whether it will help...

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
context = default   ; Default for incoming calls
allow=ilbc|gsm|ulaw|g723.1|g711
;allow=all
dtmfmode=inband
;dtmfmode=inband|rfc2833
good luck...

Girish


From: "Darren Nickerson" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
Date: Sun, 21 Dec 2003 01:29:16 -0500
Folks,

I can't seem to get DTMF signaling working properly using SJphone 
connecting
to Asterisk via a SIP connection. Here's an example of a voicemail session
where I entered 1234 for both the username and the password:

-- Incorrect password '11223344' for user '11223f344' (context = 
)

This is with dtmfmode=inband in sip.conf. With either rfc2833 or info, DTMF
tones don't seem to get 'seen' by Asterisk at all.
I'm running  CVS-12/17/03-02:39:14, in case it's relevant.

Help?

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFax Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 office
+1.215.243.8335 fax
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Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[Jim Burwell]

I had the same problem with Grandsteam phones and *.  No other hard or
soft phones have the 'double digit' problem with *.  I don't think
Asterisk can do both RFC2833 and in-band DTMF at the same time.  It
does, however, do RFC2833 and SIP Info at the same time (SIP Info
method seems to be on all the time, even when RFC2833 is selected in
the sip.conf file).  Switching the Grandsteam to SIP Info allowed it to
talk to Asterisk and fixed the double digits problem.

- Jim


Chris Albertson wrote:

  I think this is a problem on the Asterisk side.  I'm seeing
the same problem using a Grandstream Budgetone 100.  And the GS
does have setting for both in-band and RFC2833.

My guess is asterisk is accepting the DTMF tone __both__ ways
It is reading the RFC28833 stuff _and_ "hearing" the audio tones
as well.  

--- Tilghman Lesher <[EMAIL PROTECTED]> wrote:
  
  
On Sunday 21 December 2003 00:29, Darren Nickerson wrote:


  Folks,

I can't seem to get DTMF signaling working properly using SJphone
connecting to Asterisk via a SIP connection. Here's an example of a
voicemail session where I entered 1234 for both the username and
  

the


  password:

-- Incorrect password '11223344' for user '11223f344' (context
  

  
  
  
  
Changing the DTMF mode would indeed seem to be the logical
solution.  However, it appears that SJphone does not support that
option (after a quick perusal of their PDF).  You might want to file
a
bugtracker request on their website to implement that functionality.


  
  
=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

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Description: S/MIME Cryptographic Signature

Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[Chris Albertson]

I think this is a problem on the Asterisk side.  I'm seeing
the same problem using a Grandstream Budgetone 100.  And the GS
does have setting for both in-band and RFC2833.

My guess is asterisk is accepting the DTMF tone __both__ ways
It is reading the RFC28833 stuff _and_ "hearing" the audio tones
as well.  

--- Tilghman Lesher <[EMAIL PROTECTED]> wrote:
> On Sunday 21 December 2003 00:29, Darren Nickerson wrote:
> > Folks,
> >
> > I can't seem to get DTMF signaling working properly using SJphone
> > connecting to Asterisk via a SIP connection. Here's an example of a
> > voicemail session where I entered 1234 for both the username and
> the
> > password:
> >
> > -- Incorrect password '11223344' for user '11223f344' (context

> Changing the DTMF mode would indeed seem to be the logical
> solution.  However, it appears that SJphone does not support that
> option (after a quick perusal of their PDF).  You might want to file
> a
> bugtracker request on their website to implement that functionality.
>

=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

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Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[Chris Albertson]

--- Darren Nickerson <[EMAIL PROTECTED]> wrote:
> Folks,
> 
> I can't seem to get DTMF signaling working properly using SJphone
> connecting
> to Asterisk via a SIP connection. Here's an example of a voicemail
> session
> where I entered 1234 for both the username and the password:
> 
> -- Incorrect password '11223344' for user '11223f344' (context =
> )

You are lucky.  I'm getting this:

-- Incorrect password '1334' for user

When I enter "1234".  I'm using "dtmfmode=rfc2833" and a
GS Budgtone 100 phone.  Why do I getr 4x while you get 2x  ??





=
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RE: [Asterisk-Users] ivr key press?

[Joe Dennick]

Rich,
This plays into your current question and the previous one.  The 's' in
the dial plan basically applies to an inbound call that was answered by
*.  If I remember correctly, the 't' stands for any invalid entry or a
timeout waiting for DTMF digits, and I have no idea what the 'h' stands
for.  What you are doing below with Extension 620 will only come into
play IF extension 620 is dialed.  If you want your menu to apply for all
inbound callers, you need to use 's' as the extension as in the previous
example that I sent you.

The menu system is usually applied to Zap calls to the *.  The caller
(usually an outside caller) is greeted with your Auto-Attendant
greeting, and then a menu of choices like "dial 1 for support; dial 2
for sales; dial 0 for the operator, etc..."  Usually, if you want to
provide an option to dial a specific (known) extension, you would prompt
the caller to press a digit (9 for example) and then the extension;
otherwise, if your default extensions begin with '3', you would NOT use
the digit '3' as a menu option so that dialing an extension would
automatically be routed to that extension rather than the menu option
for '3'.  You might need to crank up the 'DigitTimeout' to a greater
value in order to receive all dialed digits for a specific extension.

Call me tomorrow (Sunday) and we can discuss this...

Joe

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Saturday, December 20, 2003 9:07 PM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] ivr key press?


I'm testing an ivr implementation (first time) using:
exten => 620,1,Wait,1
exten => 620,2,Answer
exten => 620,3,DigitTimeout,5
exten => 620,4,ResponseTimeout,10
exten => 620,5,Background(npi-greeting)  ; "Thanks for calling press 1
for"

exten => 1,1,Goto(npi-directory,s,1)

For initial testing, I've arbitrarily mapped this onto ext 620 (will 
change that later when things are working as expected).

The initial "npi-greeting" message essentially says "...if you know your
party's extension, you can dial it at any time.  Press 1 for Sales,
etc."

If during this initial greeting I press 3000 (which is a valid extn), I
can only press the first "3" before I get kicked out (I can't dial the
full 3000).

Am I supposed to be setting this up to expect only single-digit key
presses (instead of 3000), or am I missing something that would suggest
waiting for four key presses?



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[Asterisk-Users] SS7 API & Card Solution

[Ray Burkholder]

Title: SS7 API & Card Solution






Is this useful as a bootstrap for getting SS7 to Asterisk?


http://www.sangoma.com/api/p-api-ss7.htm


Ray Burkholder

[EMAIL PROTECTED]

http://www.oneunified.net

704 576 5101



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