date:20060621


On 6/21/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:


- Johansson Olle E [EMAIL PROTECTED] wrote:
 No. It's certainly possible but at this time there's no interaction
 between
 the RTP clients, the various channel drivers.

I believe this is incorrect; all the RTP-using channel drivers supply 
'ast_rtp_bridge' as their native bridge method, so assuming they also implement 
the 'set_rtp_peer' method, then an RTP native bridge between dissimilar 
channels should work fine. If the channel driver(s) also support sending the 
RTP peer address to the endpoint (as chan_sip does with reinvite), then a 
direct media path should also be possible.



Sorry, but I am confused.
What does this mean for the bare-bones user like me? That technically
it would be possible but that it is not implemented? (i use the sip
and h323 channels shipped with the latest sources tarball) Or that it
is possible to configure via some obscure setup file?

Thanks!

Cesc


--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] Conferencing with multiple servers


In general, you are talking of distributed conferencing, which in SIP
it was tried once to standardize but never reached anything. It is
just not commercially popular, i guess.
Now, this doesn't mean that it cannot  be done or that it has not been
done ... but it is propietary implementations. And in my particular
knowledge, i know asterisk was not the choice.

Just my 2 cents.

Cesc

On 6/20/06, Douglas Garstang [EMAIL PROTECTED] wrote:

 -Original Message-
 From: Patrick [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, June 20, 2006 12:05 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Conferencing with multiple servers


 On Tue, 2006-06-20 at 15:22 +0100, Wildheart wrote:
  Hi,
 
 I am trying to join 2 asterisk servers together using a
 sip channel.
  This is so, if a user joins a conference on box A and another user
  joins a conference on box B, providing they are in the same
 conference
  room, the two conferences are joined via the sip channel.
 We only want
  to join the conferences together if they have users in them and we
  don't want to point all the conferences to one server as we
 would like
  to try to balance the load a bit.

This is a general problem with the 'enterprise grade' aspects of Asterisk. As 
far as I know, there is no way to distribute applications (eg: Queue, Meetme 
etc) between multiple Asterisk systems. You really need to run the applications 
that will serve a common set of phones on the same Asterisk system, and then 
fail over to a secondary if necessary.

Doug.
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Re: [Asterisk-Users] Add Country to CDR's

2006-06-21 Thread Simon Woodhead

Yep, cmd setCDRUserField will do this for you assuming you have the field set up. I'd be keen to hear if anyone has a way of achieving the same thing across multiple user fields to save having to explode multiple values out of a single user field seperately.
SimonOn 6/20/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Tue, 2006-06-20 at 11:14 -0400, William Piper wrote: Thanks Bret, but how about an example or webpage? I'm not finding anything on google about this command for asterisk. What about AppendCDRUserField()... would this work?
that seems to be the same thing.the userfield lets you stick arbitrarydata into your cdr records.--Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402Utrecht NL +31 306 553058US WA +1 360 207 0479US NY +1 516 687 5200FreeWorldDialup: 635378http://www.trxtel.com
 the VoIP provider that pays you!-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.3 (GNU/Linux)iD8DBQBEmBee+1olxlzQw5cRAnnjAKCFf4NDjUlCDlf1Pb//LyeauifNbwCfUB+5cMObnxnTQYcuP3VlTYHsxZg==g7ro
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Re: [Asterisk-Users] Re: fail to make call

2006-06-21 Thread unplug


Is it the way for asterisk realtime system?
register:
UA1 --register-- asterisk1 - store user information in DB
UA2 --register-- asterisk2  store user information in DB
UA1 --invite-UA2--- asterisk1  asterisk1 query UA2 information in DB
  -asterisk1 -invite-- UA2
finally: UA1 ---asterisk1 --UA2  (2 legs keeping in asterisk1)
Am I right?


On 6/21/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:


- unplug [EMAIL PROTECTED] wrote:
   In my configuration below, I use realtime architecture in our
 system.  I have one device attached to each asterisk server.  There
 is
 no record when I issue sip show users or sip show registry in CLI.
 I
 wonder how can I know who is registered in asterisk.  What command is
 it?

The Asterisk Realtime implementation does not currently support sharing a 
registration database among multiple servers and having them be aware of the 
peers registered via the other servers. This will be addressed during the next 
development cycle, though.

--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

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Re: [Asterisk-Users] 1.2.9.1 crashed today

2006-06-21 Thread Tristan


Hi,

Here's an extract from LogWatch:

- Kernel Begin  



WARNING:  General Protection Faults in these executables
   asterisk :  1 Time(s)

-- Kernel End - 

Asterisk killed himself too after some extensives reload with the 
manager


I stopped this strange behaviour by reloading * only once a day 
automatically...



Matt a écrit :

Will try that (and a few others things) and advise when I find (if I
find) the culprit.

On 6/20/06, Steve Totaro [EMAIL PROTECTED] wrote:

Just a thought, restart the box without FOP running, and don't do any
reloads if possible.  See what happens.

Steve Totaro wrote:
 I guess that was a yes to both my scenarios, reloads and manager
 interface.
 It is still running just not responsive?
 Matt wrote:
 Arg... ok it just crashed again.  Lasted about 7 hours this time.

 On 6/20/06, Matt [EMAIL PROTECTED] wrote:
 I use FOP... I believe that that uses manager fairly extensively.

 Also... about 2-4 hours prior to the crash I had been playing around
 with getting new MOH working.. .and had reload res_musiconhold 
several

 times (5 or 6)

  Just curious, do you use the manager interface extensively?
 
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RE: [Asterisk-Users] SIPCALLID, but which callid?

2006-06-21 Thread Andreas Sikkema

 Andreas Sikkema wrote:
 
  Hi,
 
  To combine two sources of CDR's I want Asterisk to save the 
 SIP callid for
  all calls. I know there's a variable that contains the SIP 
 CallID value,
  but is this the callid value of the incoming INVITE message or the 
  outgoing
  message? Are they the same? (I've not yet checked a trace, 
 I'm sorry for
  that). I've tried to read chan_sip, but couldn't find 
 something in the 
  time
  I had today. I've found hardly any documentation o this variable, 
  apart from
  that it exists and that it contains the SIP CallID value.
 
  Can anyone enlighten me?

 They are the same on both sides.

Is this new behaviour? I've got an asterisk 1.2.5 installation that 
does not use the same CallID on both the incoming and outgoing side 
of a call through our Asterisk machine.

-- 
Andreas
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Re: [Asterisk-Users] Asterisk h323


Hi,

I am also testing asterisk with H323, with the channel included in the
latest sources. It works ( i had some problems with media
configuration when calling from an SJPhone ... but it seems more an
SJPhone problem than asterisk).
I also bridged from SIP to H323 ... it works fine.

I have a question for those out there ... i compiled with pwlib 1.9.2
and openh323 1.17.3. These are the versions mentioned in the README
file for the h323 channel.
Now, has anyone tried any newer version? I would be more comfortable
using the latest stable release for these required libraries with
asterisk ... just to make sure i get the best. Any experiences??

Cesc

On 6/20/06, Alberto Sagredo [EMAIL PROTECTED] wrote:

Im using several Asterisk Box with chanh323 from asterisk, and it works
fine.

Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability.
A fail (crash) last month with about 600 calls per day.

Regards

Alberto Sagredo


hakem voip escribió:
 You can do this by installing a h323 module.

 Conversion works simetimes good, sometimes not good. H323 behaviour on
 asterosk with my experience with kind of unpredictable.


 2006/6/20, Khaled Chehab [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]:

 Hi

 Can asterisk work as sip and h323 protocol in the same time ,and
 how is the conversion protocol works .

 Please if u know send me how to active h323 protocol or the
 conversion protocol







 Regards



 
 *
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 --
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[Asterisk-Users] IVR Applications

2006-06-21 Thread Walid Azab




Hello,
Could someone please help refer me to a 
resource where I can find material on how to write IVR applications. I am using 
[EMAIL PROTECTED] ver. 2.8.

Thanks
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[Asterisk-Users] Monitor a particular SIP call for training purposes

2006-06-21 Thread phil . dawson


Hi,

I've been asked if it is possible to
allow a user to listen in on another users call for training purposes.
I know there are ways to monitor zap channels with apps like zapscan
but I don't think this would be appropriate for these users. Can
I do:

Call comes in for user on ext 3210
User on 3222 dials a predefined number
say *888. They can then listen in on the call that user on 3210 is
having.

extensions.conf would go something like:

exten = *888,3210,ListenIn()

Is there anything like this. Basically
to allow one user to listen to another users calls ( but only that user
)


Hope I made myself clear.

Thank you in advance


Phil.
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Re: [Asterisk-Users] SIPCALLID, but which callid?

2006-06-21 Thread Erik


Andreas Sikkema wrote:

Andreas Sikkema wrote:

Hi,

To combine two sources of CDR's I want Asterisk to save the 

SIP callid for
all calls. I know there's a variable that contains the SIP 

CallID value,
but is this the callid value of the incoming INVITE message or the 
outgoing
message? Are they the same? (I've not yet checked a trace, 

I'm sorry for
that). I've tried to read chan_sip, but couldn't find 
something in the 

time
I had today. I've found hardly any documentation o this variable, 
apart from

that it exists and that it contains the SIP CallID value.

Can anyone enlighten me?



They are the same on both sides.


Is this new behaviour? I've got an asterisk 1.2.5 installation that 
does not use the same CallID on both the incoming and outgoing side 
of a call through our Asterisk machine.





SIP-CallID remains the same during a SIP Dialog, as Asterisk is a Back to Back 
UA there are 2 different call legs, so 2 different SIP-CallerID's

Erik
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RE: [Asterisk-Users] Monitor a particular SIP call for training purposes

2006-06-21 Thread Idris AVCI









Hi,



You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy.



Idris













From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, June 21, 2006
12:23 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Monitor
a particular SIP call for training purposes






Hi, 

I've
been asked if it is possible to allow a user to listen in on another users call
for training purposes. I know there are ways to monitor zap channels with
apps like zapscan but I don't think this would be appropriate for these users. Can
I do: 

Call
comes in for user on ext 3210 
User
on 3222 dials a predefined number say *888. They can then listen in on
the call that user on 3210 is having. 

extensions.conf
would go something like: 

exten
= *888,3210,ListenIn() 

Is
there anything like this. Basically to allow one user to listen to
another users calls ( but only that user ) 


Hope
I made myself clear. 

Thank
you in advance 


Phil.







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[Asterisk-Users] H.323 soft phone known to be run with asterisk.

2006-06-21 Thread Pawel

Hallo group members
Could You tell me a h.323 soft phone that runs well with asterisk.
I tried the following so far, but in general I cannot compile them (fc.3) or I 
cannot configure them to run with asterisk:

http://www.pacphone.com/downloads/PacPhoneSetup117.exe - cannot configure
http://www.sjlabs.com/softphone/SJphone-289a.exe - cannot configure
http://cphone.sourceforge.net/ - cannot compile
http://www.ekiga.org/ - cannot compile
http://www.openh323.org/ - cannot compile 

Greetings.
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Re[2]: [Asterisk-Users] Asterisk h323

2006-06-21 Thread Grigoriy Puzankin

On my Asterisk server it (chan_h323) gets 2-3 deadlocks every hour
regardless of openh323/pwlib and asterisk versions (since the
channel_h323 was not updated for a long time). The load is about 25-30
simultaneous calls (from h323 to zaptel, IAX and SIP).

I have another Asterisk server. There's about 5-7 simultaneous calls,
and deadlocks don't occur (calls go from zaptel to h323).

AS Im using several Asterisk Box with chanh323 from asterisk, and it works
AS fine.

AS Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability.
AS A fail (crash) last month with about 600 calls per day.

AS Regards

AS Alberto Sagredo


--
Grigoriy Puzankin

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[Asterisk-Users] getting zap peer of sip channel

2006-06-21 Thread Julian Lyndon-Smith


I'm wanting to capture the zap channel that a sip channel has connected to.

I came across the ${BRIDGEPEER} variable documented on the wiki, and if 
I show channel SIP/channel when a call is connected I can see 
BRIDGEPEER as one of the channel variables.


However ${BRIDGEPEER} is not set when I want it: I run a macro when the 
call has been connected.


Does anyone have a hint on how to achieve my goal ?

Many thanks

Julian
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Re: [Asterisk-Users] Re: fail to make call

2006-06-21 Thread unplug


BTW, do you mean this function will be included in next release?  When
will be the next release available?

On 6/21/06, unplug [EMAIL PROTECTED] wrote:

Is it the way for asterisk realtime system?
register:
UA1 --register-- asterisk1 - store user information in DB
UA2 --register-- asterisk2  store user information in DB
UA1 --invite-UA2--- asterisk1  asterisk1 query UA2 information in DB
   -asterisk1 -invite-- UA2
finally: UA1 ---asterisk1 --UA2  (2 legs keeping in asterisk1)
Am I right?


On 6/21/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:

 - unplug [EMAIL PROTECTED] wrote:
In my configuration below, I use realtime architecture in our
  system.  I have one device attached to each asterisk server.  There
  is
  no record when I issue sip show users or sip show registry in CLI.
  I
  wonder how can I know who is registered in asterisk.  What command is
  it?

 The Asterisk Realtime implementation does not currently support sharing a 
registration database among multiple servers and having them be aware of the peers 
registered via the other servers. This will be addressed during the next 
development cycle, though.

 --
 Kevin P. Fleming
 Senior Software Engineer
 Digium, Inc.

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Re: [Asterisk-Users] Monitor a particular SIP call for training purposes

2006-06-21 Thread Steve Totaro

Anyone know why with Chanspy if I dial a specific extension and press # 
I get a random agent and not the one I dialed?


Idris AVCI wrote:


Hi,

 

You can try ChanSpy 
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy.


 


Idris

 

 




*From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
*Sent:* Wednesday, June 21, 2006 12:23 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Monitor a particular SIP call for training 
purposes


 



Hi,

I've been asked if it is possible to allow a user to listen in on 
another users call for training purposes.  I know there are ways to 
monitor zap channels with apps like zapscan but I don't think this 
would be appropriate for these users.  Can I do:


Call comes in for user on ext 3210
User on 3222 dials a predefined number say *888.  They can then listen 
in on the call that user on 3210 is having.


extensions.conf would go something like:

exten = *888,3210,ListenIn()

Is there anything like this.  Basically to allow one user to listen to 
another users calls ( but only that user )



Hope I made myself clear.

Thank you in advance


Phil.



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Re: [Asterisk-Users] getting zap peer of sip channel (solved)

2006-06-21 Thread Julian Lyndon-Smith


Doh, I am so stupid.

The macro is executing in the zap channel !

Julian.

Julian Lyndon-Smith wrote:

I'm wanting to capture the zap channel that a sip channel has connected to.

I came across the ${BRIDGEPEER} variable documented on the wiki, and if 
I show channel SIP/channel when a call is connected I can see 
BRIDGEPEER as one of the channel variables.


However ${BRIDGEPEER} is not set when I want it: I run a macro when the 
call has been connected.


Does anyone have a hint on how to achieve my goal ?

Many thanks

Julian
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Re: [Asterisk-Users] 1.2.9.1 crashed today


Ok,
So here's some information I've, to this point, left out.   I applied
the patch to allow * to be pressed in queues to park a call (*270).
After reverting the patch the system seems stable.  So, it almost
seems like the crash is directly related to the number of times
someone parks a call that came in a queue.

I don't understand, though

  /* terminates call */
ast_frfree(f);
f = NULL;

How does removing those 2 lines cause asterisk to crash?   Basically
the code looks for a *, if it sees it it runs that... so the patch
removes those lines.. so when asterisk sees a * it doesn't do
anything.The variables are so cryptic I can't exactly figure out
what its doing.. is ast_frfree(f) suppose to 'free' something, perhaps
that is not getting freed?  But then what is f = NULL.  Yikes, who
comes up with these variable names?

On 6/21/06, Tristan [EMAIL PROTECTED] wrote:

Hi,

Here's an extract from LogWatch:

 - Kernel Begin 


 WARNING:  General Protection Faults in these executables
asterisk :  1 Time(s)

 -- Kernel End -

Asterisk killed himself too after some extensives reload with the
manager

I stopped this strange behaviour by reloading * only once a day
automatically...


Matt a écrit :
 Will try that (and a few others things) and advise when I find (if I
 find) the culprit.

 On 6/20/06, Steve Totaro [EMAIL PROTECTED] wrote:
 Just a thought, restart the box without FOP running, and don't do any
 reloads if possible.  See what happens.

 Steve Totaro wrote:
  I guess that was a yes to both my scenarios, reloads and manager
  interface.
  It is still running just not responsive?
  Matt wrote:
  Arg... ok it just crashed again.  Lasted about 7 hours this time.
 
  On 6/20/06, Matt [EMAIL PROTECTED] wrote:
  I use FOP... I believe that that uses manager fairly extensively.
 
  Also... about 2-4 hours prior to the crash I had been playing around
  with getting new MOH working.. .and had reload res_musiconhold
 several
  times (5 or 6)
 
   Just curious, do you use the manager interface extensively?
  
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[Asterisk-Users] SPA-2002 call HANGUP. May be a SIP bug.

2006-06-21 Thread Dmytro Mishchenko

Hello,
We have problems with Asterisk and Sipura SPA-2002. 
SPA is behind the NAT. Asterisk has nat=yes.

Sometimes call doesn't hangup when user finish the call and hangup the 
headset.

In this case during all conversation SIP packets contains 
Call-ID: [EMAIL PROTECTED] 
but the final BYE packet from adapter contains 
Call-ID: [EMAIL PROTECTED]
Is such scenario correct from SIP protocol point of view?

After receiving such packet Asterisk trying to destroy:
Destroying call '[EMAIL PROTECTED]'
But I suppose that such call doesn't exists and call remains active.

Thanks,
Dmitry.
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Re: [Asterisk-Users] H.323 soft phone known to be run with asterisk.


I had problems with sjphone ... same version as yours.
Finally, i managed to solve it by:
- in sjphone, media channels settings: untick Use remote codec
preferences and Open audio streams after remote opened ... it was
trial-error ... now it works (to Echo and Sip-H323 call).
- in asterisk, h323.conf ... the codec configuration ... i commented
all lines related to it ...
;disallow=all
;allow=all
;allow=gsm
;disallow=g723.1
(just in case)
(again, trial-error)

Cesc

On 6/21/06, Pawel [EMAIL PROTECTED] wrote:

Hallo group members
Could You tell me a h.323 soft phone that runs well with asterisk.
I tried the following so far, but in general I cannot compile them (fc.3) or I 
cannot configure them to run with asterisk:

http://www.pacphone.com/downloads/PacPhoneSetup117.exe - cannot configure
http://www.sjlabs.com/softphone/SJphone-289a.exe - cannot configure
http://cphone.sourceforge.net/ - cannot compile
http://www.ekiga.org/ - cannot compile
http://www.openh323.org/ - cannot compile

Greetings.
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Re: [Asterisk-Users] 1.2.9.1 crashed today

2006-06-21 Thread BJ Weschke


On 6/21/06, Matt [EMAIL PROTECTED] wrote:

Ok,
So here's some information I've, to this point, left out.   I applied
the patch to allow * to be pressed in queues to park a call (*270).
After reverting the patch the system seems stable.  So, it almost
seems like the crash is directly related to the number of times
someone parks a call that came in a queue.

I don't understand, though

   /* terminates call */
 ast_frfree(f);
 f = NULL;

How does removing those 2 lines cause asterisk to crash?   Basically
the code looks for a *, if it sees it it runs that... so the patch
removes those lines.. so when asterisk sees a * it doesn't do
anything.The variables are so cryptic I can't exactly figure out
what its doing.. is ast_frfree(f) suppose to 'free' something, perhaps
that is not getting freed?  But then what is f = NULL.  Yikes, who
comes up with these variable names?



Yes, ast_frfree(..) is supposed to free any allocated memory
associated with the pointer to the Asterisk frame structure given to
it. If the structure has already been freed somewhere else in the code
and you try to call this again, you will segfault Asterisk. If you
don't call it at all, you will leak memory associated with the
structure. The pointer is being set to NULL after the call so that
further checks of the pointer itself will be able to determine that
the structure has been free'd already and doesn't need to be done
again (preventing a segfault from calling it twice).

--
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[Asterisk-Users] zapata.conf: recent changes?

Hi,
after a few of upgrades, I noticed these messages in full debug log:

Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring internationalprefix
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring nationalprefix
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring localprefix
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring privateprefix
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring unknownprefix
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring priindication
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring facilityenable
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling

A lot of changes?

DV

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[Asterisk-Users] database space

2006-06-21 Thread Khaled Chehab











Dear 

I am using [EMAIL PROTECTED] , and I have 2 hard disks on the
system ,how can I put the database (CDR) on the second hard disk . 





Regards

M. Khaled
Chehab

Monitoring  Operationg Engineer 

Xplorium

Tel: +961 1
868686

Fax: +961 1
808810

e-mail: [EMAIL PROTECTED]








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Re: [Asterisk-Users] 1.2.9.1 crashed today


On 6/21/06, BJ Weschke [EMAIL PROTECTED] wrote:

On 6/21/06, Matt [EMAIL PROTECTED] wrote:
 Ok,
 So here's some information I've, to this point, left out.   I applied
 the patch to allow * to be pressed in queues to park a call (*270).
 After reverting the patch the system seems stable.  So, it almost
 seems like the crash is directly related to the number of times
 someone parks a call that came in a queue.

 I don't understand, though

/* terminates call */
  ast_frfree(f);
  f = NULL;

 How does removing those 2 lines cause asterisk to crash?   Basically
 the code looks for a *, if it sees it it runs that... so the patch
 removes those lines.. so when asterisk sees a * it doesn't do
 anything.The variables are so cryptic I can't exactly figure out
 what its doing.. is ast_frfree(f) suppose to 'free' something, perhaps
 that is not getting freed?  But then what is f = NULL.  Yikes, who
 comes up with these variable names?


 Yes, ast_frfree(..) is supposed to free any allocated memory
associated with the pointer to the Asterisk frame structure given to
it. If the structure has already been freed somewhere else in the code
and you try to call this again, you will segfault Asterisk. If you
don't call it at all, you will leak memory associated with the
structure. The pointer is being set to NULL after the call so that
further checks of the pointer itself will be able to determine that
the structure has been free'd already and doesn't need to be done
again (preventing a segfault from calling it twice).



Ok... that makes sence, and is probably why asterisk was crashing then
(good news is since reversing the patch and installing clean 1.2.9.1
we've been rock solid over night).   Now.. my question is... how do
those 2 lines of code disconnect a call?
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Re: [Asterisk-Users] zapata.conf: recent changes?

2006-06-21 Thread Tzafrir Cohen

On Wed, Jun 21, 2006 at 01:57:36PM +0200, Mimmus wrote:
 Hi,
 after a few of upgrades, I noticed these messages in full debug log:
 
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring internationalprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring nationalprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring localprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring privateprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring unknownprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring priindication
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring facilityenable
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling

Missing '[channels]' ?

Could you provide the complete file?

-- 
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406   
[EMAIL PROTECTED]  http://www.xorcom.com
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[Asterisk-Users] syntax error

Does anyone know why this row:

 exten = s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?4:3)

generate this error:

ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_NE,
expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
 != 
 ^

?

I was unable to debug it.
--
DV

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[Asterisk-Users] USB handset options for softphones

2006-06-21 Thread klubarpop

Are there any traditional telephone set-looking handset for use with a
softphone?  All the options I've found are the headset type.  I'm looking
for something more traditional--it should look like a small deskset,
cellphone or cordless phone, perhaps with a dial pad and a couple of buttons
that integrates well with a softphone client.  My users don't want to pick
up a headset/mic when the phone rings or use the built-in computer
speaker/mic.  

I've looked at the Clarisys i750H, but it's not yet shipping.  Any other
popular options? And softphone clients do you use with them?

Many thanks
Ken lubar

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Re: [Asterisk-Users] Upgrading asterisk

2006-06-21 Thread Thomas Kenyon

Thomas Kenyon wrote:
 Doug Lytle wrote:
   
 Thomas Kenyon wrote:
 
 Is it neccesary to upgrade Zaptel at the same time as upgrading
 asterisk.

   
   
 I do as a matter or course.  Libpri, Zaptel, Asterisk, Asterisk-addons
 and Sounds.

 Doug

 
 The problem with zaptel is that even if you can unload the modules and
 reload them again, it still involves some downtime.

 Will look at doing it over the weekend (unless I get another crash.)
   
With new zaptel, this is still happening, still can't find anything in
the log files that is relevant, it just seems to be business as usual
then freeze.

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Re: [Asterisk-Users] 1.2.9.1 crashed today

2006-06-21 Thread BJ Weschke


Do you have that patch/commit message from svn-commits where this
code was introduced so we can track it back?



On 6/21/06, Matt [EMAIL PROTECTED] wrote:

On 6/21/06, BJ Weschke [EMAIL PROTECTED] wrote:
 On 6/21/06, Matt [EMAIL PROTECTED] wrote:
  Ok,
  So here's some information I've, to this point, left out.   I applied
  the patch to allow * to be pressed in queues to park a call (*270).
  After reverting the patch the system seems stable.  So, it almost
  seems like the crash is directly related to the number of times
  someone parks a call that came in a queue.
 
  I don't understand, though
 
 /* terminates call */
   ast_frfree(f);
   f = NULL;
 
  How does removing those 2 lines cause asterisk to crash?   Basically
  the code looks for a *, if it sees it it runs that... so the patch
  removes those lines.. so when asterisk sees a * it doesn't do
  anything.The variables are so cryptic I can't exactly figure out
  what its doing.. is ast_frfree(f) suppose to 'free' something, perhaps
  that is not getting freed?  But then what is f = NULL.  Yikes, who
  comes up with these variable names?
 

  Yes, ast_frfree(..) is supposed to free any allocated memory
 associated with the pointer to the Asterisk frame structure given to
 it. If the structure has already been freed somewhere else in the code
 and you try to call this again, you will segfault Asterisk. If you
 don't call it at all, you will leak memory associated with the
 structure. The pointer is being set to NULL after the call so that
 further checks of the pointer itself will be able to determine that
 the structure has been free'd already and doesn't need to be done
 again (preventing a segfault from calling it twice).


Ok... that makes sence, and is probably why asterisk was crashing then
(good news is since reversing the patch and installing clean 1.2.9.1
we've been rock solid over night).   Now.. my question is... how do
those 2 lines of code disconnect a call?
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AW: [Asterisk-Users] syntax error

2006-06-21 Thread Marc Rohlfing

  Hi,

 Does anyone know why this row:
  exten = s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
 ${RGPREFIX}]?4:3)

took me some squinting, but the parantheses seem correct - so I presume
the Asterisk parser can't cope with that convoluted an expression (using
a function within a variable, basically).
Try putting LEN(${RGPREFIX}) into a separate variable first, then refer
to it in your GotoIf in a second statement.

  Marc

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RE: [Asterisk-Users] zapata.conf: recent changes?

 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Tzafrir Cohen
 
 On Wed, Jun 21, 2006 at 01:57:36PM +0200, Mimmus wrote:
  Hi,
  after a few of upgrades, I noticed these messages in full debug log:
  
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring 
 switchtype Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring 
 prilocaldialplan Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring 
 internationalprefix Jun 
  21 12:58:11 WARNING[27273] chan_zap.c: Ignoring 
 nationalprefix Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring localprefix Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring privateprefix Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring unknownprefix Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring priindication Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring facilityenable Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling Jun 21 
  12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling
 
 Missing '[channels]' ?
 
 Could you provide the complete file?

;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=it
switchtype=euroisdn
pridialplan=dynamic
prilocaldialplan=local
overlapdial=yes

internationalprefix=00
nationalprefix=0
localprefix=0984
privateprefix=0984899
unknownprefix=

priindication=inband
facilityenable=yes
rxgain=0.0
txgain=0.0
jitterbuffers=2
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
usecallingpres=yes
useincomingcalleridonzaptransfer=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
immediate=no
callerid=asreceived
musiconhold=native-random

; Incoming only
group=0
signalling=pri_cpe
context=from-pstn
faxdetect=no
channel = 1-10

; Outgoing (only?)
group=1
faxdetect=no
channel = 11-15,17-21

; Not used
group=3
faxdetect=no
channel = 22-31

; To/From Alcatel
group=2
signalling=pri_net
faxdetect=no
context=from-alcatel
channel = 32-46,48-62



Thanks
DV

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Re: [Asterisk-Users] show queue ... Invalid

2006-06-21 Thread Denis Shaposhnikov

 Kevin == Kevin P Fleming [EMAIL PROTECTED] writes:

 Kevin I've just reviewed the code and this should be working
 Kevin properly... please do a 'set debug 3' and enable the 'debug'

I've found the problem. That's because I've loaded app_queue.so before
chan_sip.so in modules.conf.

-- 
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xmpp:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://neva.vlink.ru/~dsh/
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Re: [Asterisk-Users] 1.2.9.1 crashed today


It fuzzed on the first chunk, so I wiped the source directory, re
un-tarred asterisk so it was clean, and then manually applied the
changes.. doing removes and adds where necessary.  So, unfortunately
no, I do not have the information, though I suppose I could run it
again.

Keep in mind, also, that I'm running 1.2.9.1, not svn-trunk.

On 6/21/06, BJ Weschke [EMAIL PROTECTED] wrote:

 Do you have that patch/commit message from svn-commits where this
code was introduced so we can track it back?



On 6/21/06, Matt [EMAIL PROTECTED] wrote:
 On 6/21/06, BJ Weschke [EMAIL PROTECTED] wrote:
  On 6/21/06, Matt [EMAIL PROTECTED] wrote:
   Ok,
   So here's some information I've, to this point, left out.   I applied
   the patch to allow * to be pressed in queues to park a call (*270).
   After reverting the patch the system seems stable.  So, it almost
   seems like the crash is directly related to the number of times
   someone parks a call that came in a queue.
  
   I don't understand, though
  
  /* terminates call */
ast_frfree(f);
f = NULL;
  
   How does removing those 2 lines cause asterisk to crash?   Basically
   the code looks for a *, if it sees it it runs that... so the patch
   removes those lines.. so when asterisk sees a * it doesn't do
   anything.The variables are so cryptic I can't exactly figure out
   what its doing.. is ast_frfree(f) suppose to 'free' something, perhaps
   that is not getting freed?  But then what is f = NULL.  Yikes, who
   comes up with these variable names?
  
 
   Yes, ast_frfree(..) is supposed to free any allocated memory
  associated with the pointer to the Asterisk frame structure given to
  it. If the structure has already been freed somewhere else in the code
  and you try to call this again, you will segfault Asterisk. If you
  don't call it at all, you will leak memory associated with the
  structure. The pointer is being set to NULL after the call so that
  further checks of the pointer itself will be able to determine that
  the structure has been free'd already and doesn't need to be done
  again (preventing a segfault from calling it twice).
 

 Ok... that makes sence, and is probably why asterisk was crashing then
 (good news is since reversing the patch and installing clean 1.2.9.1
 we've been rock solid over night).   Now.. my question is... how do
 those 2 lines of code disconnect a call?
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[Asterisk-Users] database copy in asterisk

2006-06-21 Thread Shenen Shenen

Hi!I've 2 asteriskAtHome;
How can I copy one database where are put all the sip authentificated registration to another one database on one other asteriskAthome so I've always the same Sip registrated and if one linux falls down I can run the other one without problems? 

Which files must I copy?then..I'll use a ssh scritp for this, I want only know which filesI must copy...

100 thanks.
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[Asterisk-Users] Asterisk 1.2.7/9.1 mp3 volume is good, wav file of same volume are too loud!


Ok,
Here's another bizarre one (no strange curve balls to throw this time :P).

I have several mp3 files of some easy listening music that I pulled
off some CDs we have.  They sounds fine and are at a nice volume
level.

When I run this script I wrote:
(I run it by doing ./script  filename)
mpg123 -s --rate 44100 --mono $1.mp3  $1.raw
sox -r 44100 -w -s -c 1 $1.raw -r 8000 -c 1 $1.wav
rm $1.raw

The wav files comes out with the same volume (I checked using a
program called goldwave)... and infact if I play it on my laptop it
sounds the same and the VU meters show the same volume
characteristicshowever, when I place someone on hold, the music
becomes deafening.. ok maybe not that bad, but it's very much louder
to the point of almost being too loud.

Is there something volume wise different about the internal mp3 and wav players?
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[Asterisk-Users] Asterisk h323

2006-06-21 Thread Khaled Chehab


 Hi

 How Can asterisk work as sip and h323 protocol in the same time ,and how
is  
 the
 conversion protocol works .

 Please if u know send me how to active h323 protocol or the conversion
 protocol




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electronic message do not necessarily reflect views of Xplorium or its 
subsidiaries and associates.

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Re: [Asterisk-Users] zapata.conf: recent changes?

2006-06-21 Thread Wolfgang Zweimueller

Mimmus [EMAIL PROTECTED] writes:

 Hi,
 after a few of upgrades, I noticed these messages in full debug log:

 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring internationalprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring nationalprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring localprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring privateprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring unknownprefix
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring priindication
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring facilityenable
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling
 Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling

 A lot of changes?

No, IMHO does it appear when you issue a reload command on the
CLI. Because this options need a complete *-restart.


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




Voip Supply has a number of USB handsets available... see http://www.voipsupply.com/index.php?cPath=95_258 I'm not affiliated with them beyond being a happy customer.



There are also a few USB handsets that look like desk phones. I have seen these offered in the $50-70 range, some even with speaker phones. Not sure where though. Try a quick search on Ebay.



Michael





On Wed, 21 Jun 2006 08:19:44 -0400, klubarpop wrote:



Are there any traditional telephone set-looking handset for use with a

softphone?  All the options I've found are the headset type.  I'm looking

for something more traditional--it should look like a small deskset,

cellphone or cordless phone, perhaps with a dial pad and a couple of buttons

that integrates well with a softphone client.  My users don't want to pick

up a headset/mic when the phone rings or use the built-in computer

speaker/mic.  



I've looked at the Clarisys i750H, but it's not yet shipping.  Any other

popular options? And softphone clients do you use with them?



Many thanks

Ken lubar



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




A quick Google returned this USB desk phone for $32http://www.inkjetcartridge.com/deskphone.html.



Michael



On Wed, 21 Jun 2006 08:19:44 -0400, klubarpop wrote:



Are there any traditional telephone set-looking handset for use with a

softphone?  All the options I've found are the headset type.  I'm looking

for something more traditional--it should look like a small deskset,

cellphone or cordless phone, perhaps with a dial pad and a couple of buttons

that integrates well with a softphone client.  My users don't want to pick

up a headset/mic when the phone rings or use the built-in computer

speaker/mic.  



I've looked at the Clarisys i750H, but it's not yet shipping.  Any other

popular options? And softphone clients do you use with them?



Many thanks

Ken lubar



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Re: [Asterisk-Users] Anyone using VoIP WiFi phones?

2006-06-21 Thread Andrew Kohlsmith

On Tuesday 20 June 2006 17:28, Karl J. Vesterling wrote:
 I could care less about the color LCD, or browsing the web on the
 phone, but a good speakerphone built into a cordless WiFi is
 definitely a requirement that needs to be met before I purchase one.

Agreed, although every single manufacturer is missing out on Bluetooth.  This 
is a requirement for us.  Corded headsets are such a pain in the ass, 
especially on cordless phones.

-A.
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Re: [Asterisk-Users] database copy in asterisk


Are you asking about the internal Asterisk database?   Or the
configuration files?

You don't need to copy anything about sip registration.   Just either
copy the .conf files from /etc/asterisk (so copy all of /etc/asterisk
over).  Or copy the MySQL database over and have it recreate the
files.

On 6/21/06, Shenen Shenen [EMAIL PROTECTED] wrote:


Hi!I've 2 asteriskAtHome;
How can I copy one database where are put all the sip authentificated
registration to another one database on one other asteriskAthome so I've
always the same Sip registrated and if one linux falls down I can run the
other one without problems?
Which files must I copy?then..I'll use a ssh scritp for this, I want only
know which files I must copy...

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

2006-06-21 Thread Steven

We use the AU-100 from (some Chinese company)
They work OK.
Our only issue is no having an external ringer.

We originally went with these because our Dell soundcards made the microphone 
input sound awful.
USB sound devices have their own sound card on board.

-- 
-- 
Steven

http://www.glimasoutheast.org



klubarpop [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
 Are there any traditional telephone set-looking handset for use with a
 softphone?  All the options I've found are the headset type.  I'm looking
 for something more traditional--it should look like a small deskset,
 cellphone or cordless phone, perhaps with a dial pad and a couple of buttons
 that integrates well with a softphone client.  My users don't want to pick
 up a headset/mic when the phone rings or use the built-in computer
 speaker/mic.

 I've looked at the Clarisys i750H, but it's not yet shipping.  Any other
 popular options? And softphone clients do you use with them?

 Many thanks
 Ken lubar

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RE: [Asterisk-Users] zapata.conf: recent changes?


 No, IMHO does it appear when you issue a reload command on 
 the CLI. Because this options need a complete *-restart.
Yes, they appears when I issue a reload.
I will check if there are also when I restart.

Thanks
DV

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Re: [Asterisk-Users] database copy in asterisk

2006-06-21 Thread Time Bandit


Which files must I copy?then..I'll use a ssh scritp for this, I want only
know which files I must copy...

the MySQL files are usually in /var/lib/mysql. The databse you want
to copy is asterisk

hth
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[Asterisk-Users] CDRTool

2006-06-21 Thread Giedrius Augys

Has anybody patched sucessfully cdr_addon_mysql.c , cause I get error . And does work cdrtool web interface for you? Maybe you can give me some advices
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Re: [Asterisk-Users] Anyone using VoIP WiFi phones?

2006-06-21 Thread Cory Andrews

MVox makes a BlueTooth enabled speakerphone that works nicely with 
softphones, but is not inexpensive.  Details can be found here 
http://www.mvox.com/


Cory J Andrews

VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
- Original Message - 
From: Andrew Kohlsmith [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Wednesday, June 21, 2006 9:38 AM
Subject: Re: [Asterisk-Users] Anyone using VoIP WiFi phones?



On Tuesday 20 June 2006 17:28, Karl J. Vesterling wrote:

I could care less about the color LCD, or browsing the web on the
phone, but a good speakerphone built into a cordless WiFi is
definitely a requirement that needs to be met before I purchase one.


Agreed, although every single manufacturer is missing out on Bluetooth. 
This

is a requirement for us.  Corded headsets are such a pain in the ass,
especially on cordless phones.

-A.
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[Asterisk-Users] MySQL Realtime Voicemail Connection Lost

2006-06-21 Thread Douglas Garstang

I'm using realtime for voicemail users, and for reasons that I don't yet 
understand, when it doesn't get used for a while (like overnight), the first 
connection attempt of the day will display this on the console.

Jun 21 07:54:00 ERROR[8112]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: 
Unknown connection error: (2006) MySQL server has gone away
Jun 21 07:54:01 NOTICE[8120]: rtp.c:564 ast_rtp_read: Unknown RTP codec 96 
received
-- Executing VoiceMail(SIP/xxx.187.142.186-b773c428, [EMAIL PROTECTED]) 
in new stack
Jun 21 07:54:01 ERROR[8120]: res_config_mysql.c:623 mysql_reconnect: MySQL 
RealTime: Failed to reconnect. Check debug for more info.
Jun 21 07:54:01 WARNING[8120]: app_voicemail.c:2411 leave_voicemail: No entry 
in voicemail config file for '2944017'

The next connection attempt will work. Happens like clockwork every morning. It 
would seem that Asterisk is not reconnecting the first time, even when it says 
it is. I'm thinking I may open a bug on this. Has anyone else encountered this 
behaviour?

Doug.

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FW: [Asterisk-Users] syntax error

2006-06-21 Thread Rob Thomas

(Try again from the proper email address)

--Rob


-Original Message-
From: Rob Thomas 
Sent: Thursday, 22 June 2006 12:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] syntax error


That's freePBX or AMP code that we've since fixed - The replacement line is

exten = s,2,GotoIf($[${CALLERID(name):0:${LEN(${RGPREFIX})}} != 
${RGPREFIX}]?4:3)  ; check for old prefix

(Upgrade to freePBX 2.1.1, it's much better, really)

--Rob
(freePBX dev)

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, 21 June 2006 10:17 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] syntax error
 
 
 Does anyone know why this row:
 
  exten = s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
 ${RGPREFIX}]?4:3)
 
 generate this error:
 
 ast_expr2.fl: ast_yyerror(): syntax error: syntax error,
 unexpected TOK_NE, expecting TOK_MINUS or TOK_COMPL or TOK_LP 
 or TOKEN; Input:  != 
  ^
 
 ?
 
 I was unable to debug it.
 --
 DV
 
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FW: [Asterisk-Users] syntax error

2006-06-21 Thread Rob Thomas

Third time's the charm.. (Email server is sending from wrong address!)

--Rob

-Original Message-
From: Rob Thomas 
Sent: Thursday, 22 June 2006 12:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] syntax error


That's freePBX or AMP code that we've since fixed - The replacement line is

exten = s,2,GotoIf($[${CALLERID(name):0:${LEN(${RGPREFIX})}} != 
${RGPREFIX}]?4:3)  ; check for old prefix

(Upgrade to freePBX 2.1.1, it's much better, really)

--Rob
(freePBX dev)

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, 21 June 2006 10:17 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] syntax error
 
 
 Does anyone know why this row:
 
  exten = s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
 ${RGPREFIX}]?4:3)
 
 generate this error:
 
 ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected 
 TOK_NE, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:  
 !=  ^
 
 ?
 
 I was unable to debug it.
 --
 DV
 
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FW: [Asterisk-Users] zapata.conf: recent changes?

2006-06-21 Thread Rob Thomas

And I'll resend this one too. Silly scalix.

--Rob

-Original Message-
From: Rob Thomas 
Sent: Thursday, 22 June 2006 12:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] zapata.conf: recent changes?


Looks like you've stopped compiling libpri. All those options that are being 
ignored, are being ignored because they're for PRI, and you don't have PRI 
support in zaptel.

--Rob


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, 21 June 2006 10:48 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] zapata.conf: recent changes?
 
 
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Tzafrir Cohen
  
  On Wed, Jun 21, 2006 at 01:57:36PM +0200, Mimmus wrote:
   Hi,
   after a few of upgrades, I noticed these messages in full
 debug log:
   
   Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring
  switchtype Jun 21
   12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan Jun 21 
   12:58:11 WARNING[27273] chan_zap.c: Ignoring
  prilocaldialplan Jun 21
   12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial Jun 21 
   12:58:11 WARNING[27273] chan_zap.c: Ignoring
  internationalprefix Jun
   21 12:58:11 WARNING[27273] chan_zap.c: Ignoring
  nationalprefix Jun 21
   12:58:11 WARNING[27273] chan_zap.c: Ignoring localprefix Jun 21 
   12:58:11 WARNING[27273] chan_zap.c: Ignoring privateprefix Jun 21 
   12:58:11 WARNING[27273] chan_zap.c: Ignoring unknownprefix Jun 21 
   12:58:11 WARNING[27273] chan_zap.c: Ignoring priindication Jun 21 
   12:58:11 WARNING[27273] chan_zap.c: Ignoring
 facilityenable Jun 21
   12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling Jun 21
   12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling
  
  Missing '[channels]' ?
  
  Could you provide the complete file?
 
 ;
 ; Zapata telephony interface
 ;
 ; Configuration file
 
 [trunkgroups]
 
 [channels]
 
 language=it
 switchtype=euroisdn
 pridialplan=dynamic
 prilocaldialplan=local
 overlapdial=yes
 
 internationalprefix=00
 nationalprefix=0
 localprefix=0984
 privateprefix=0984899
 unknownprefix=
 
 priindication=inband
 facilityenable=yes
 rxgain=0.0
 txgain=0.0
 jitterbuffers=2
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 callwaitingcallerid=yes
 usecallingpres=yes
 useincomingcalleridonzaptransfer=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=yes
 echotraining=800
 immediate=no
 callerid=asreceived
 musiconhold=native-random
 
 ; Incoming only
 group=0
 signalling=pri_cpe
 context=from-pstn
 faxdetect=no
 channel = 1-10
 
 ; Outgoing (only?)
 group=1
 faxdetect=no
 channel = 11-15,17-21
 
 ; Not used
 group=3
 faxdetect=no
 channel = 22-31
 
 ; To/From Alcatel
 group=2
 signalling=pri_net
 faxdetect=no
 context=from-alcatel
 channel = 32-46,48-62
 
 
 
 Thanks
 DV
 
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Re: [Asterisk-Users] zapata.conf: recent changes?

2006-06-21 Thread Tzafrir Cohen

On Wed, Jun 21, 2006 at 04:06:07PM +0200, Mimmus wrote:
 
  No, IMHO does it appear when you issue a reload command on 
  the CLI. Because this options need a complete *-restart.
 Yes, they appears when I issue a reload.
 I will check if there are also when I restart.

Restart? Who needs a restart?

http://bugs.digium.com/view.php?id=6955

The .dpatch there should be for 1.2 . If there's any problem with it,
let me know and I'll post the copy I currently use for my deb.

-- 
Tzafrir Cohen  sip:[EMAIL PROTECTED]
icq#16849755   iax:[EMAIL PROTECTED]
+972-50-7952406   
[EMAIL PROTECTED]  http://www.xorcom.com
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[Asterisk-Users] forward a call to a SIP account on a remote server

2006-06-21 Thread nik600


Hi

i will forward a call to a remote server (only for one account)

is this sintax correct?

exten = 33347563,1,Dial(SIP/[EMAIL PROTECTED])

thanks
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Re[2]: [Asterisk-Users] H.323 soft phone known to be run with asterisk.

2006-06-21 Thread Tigran Kocharyan

Try Ekiga,
It works as and is stabil as well.


Wednesday, June 21, 2006, 1:53:33 PM, you wrote:

 I had problems with sjphone ... same version as yours.
 Finally, i managed to solve it by:
 - in sjphone, media channels settings: untick Use remote codec
 preferences and Open audio streams after remote opened ... it was
 trial-error ... now it works (to Echo and Sip-H323 call).
 - in asterisk, h323.conf ... the codec configuration ... i commented
 all lines related to it ...
 ;disallow=all
 ;allow=all
 ;allow=gsm
 ;disallow=g723.1
 (just in case)
 (again, trial-error)

 Cesc

 On 6/21/06, Pawel [EMAIL PROTECTED] wrote:
 Hallo group members
 Could You tell me a h.323 soft phone that runs well with asterisk.
 I tried the following so far, but in general I cannot compile
 them (fc.3) or I cannot configure them to run with asterisk:

 http://www.pacphone.com/downloads/PacPhoneSetup117.exe - cannot configure
 http://www.sjlabs.com/softphone/SJphone-289a.exe - cannot configure
 http://cphone.sourceforge.net/ - cannot compile
 http://www.ekiga.org/ - cannot compile
 http://www.openh323.org/ - cannot compile

 Greetings.
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-- 
Best regards,
 Tigranmailto:[EMAIL PROTECTED]


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[Asterisk-Users] Telsey CPV

2006-06-21 Thread Morten Isaksen

Hi!

Has anyone used this box together with Asterisk?

I have a hard time finding information about this product. I have no manual and Telseys support does not answer any e-mails and you cannot download them on their homepage. 

If anyone one has any information about configuration via tftp I would really appreciate it.-- Morten Isaksenhttp://www.misak.dk/blog/ 
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Re: [Asterisk-Users] Anyone using VoIP WiFi phones?




Sorry to be contrarian..but I bought the MV900 (from you actually) and I'm not really impressed with it. Prior to it I had been using the Phoenix Audio Duet which is a vastly superior device, albeit lacking the bluetooth capability.



As soon as the new Polycom Communicator is available I expect I'll try that and pawn the MV900 off on some unsuspecting colleague.



Michael



On Wed, 21 Jun 2006 10:25:41 -0400, Cory Andrews wrote:



MVox makes a BlueTooth enabled speakerphone that works nicely with 

softphones, but is not inexpensive.  Details can be found here 

http://www.mvox.com/



Cory J Andrews



VOIPSupply.com

454 Sonwil Drive

Buffalo, NY 14225

++

voice - 716.630.1555 X22

email - [EMAIL PROTECTED]

AIM - B2CORY

- Original Message - 

From: "Andrew Kohlsmith" [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com

Sent: Wednesday, June 21, 2006 9:38 AM

Subject: Re: [Asterisk-Users] Anyone using VoIP WiFi phones?





 On Tuesday 20 June 2006 17:28, Karl J. Vesterling wrote:

 I could care less about the color LCD, or browsing the web on the

 phone, but a good speakerphone built into a cordless WiFi is

 definitely a requirement that needs to be met before I purchase one.



 Agreed, although every single manufacturer is missing out on Bluetooth. 

 This

 is a requirement for us.  Corded headsets are such a pain in the ass,

 especially on cordless phones.



 -A.

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Re: [Asterisk-Users] H.323 soft phone known to be run with asterisk.

2006-06-21 Thread Pawel

Hallo Cesc
Cesc writes:
  I had problems with sjphone ... same version as yours.
  Finally, i managed to solve it by:
  - in sjphone, media channels settings: untick Use remote codec
  preferences and Open audio streams after remote opened ... it was
  trial-error ... now it works (to Echo and Sip-H323 call).
  - in asterisk, h323.conf ... the codec configuration ... i commented
  all lines related to it ...
  ;disallow=all
  ;allow=all
  ;allow=gsm
  ;disallow=g723.1
  (just in case)
  (again, trial-error)
  
  Cesc
  
  On 6/21/06, Pawel [EMAIL PROTECTED] wrote:
   Hallo group members
   Could You tell me a h.323 soft phone that runs well with asterisk.
   I tried the following so far, but in general I cannot compile them (fc.3) 
   or I cannot configure them to run with asterisk:
  
   http://www.pacphone.com/downloads/PacPhoneSetup117.exe - cannot configure
   http://www.sjlabs.com/softphone/SJphone-289a.exe - cannot configure
   http://cphone.sourceforge.net/ - cannot compile
   http://www.ekiga.org/ - cannot compile
   http://www.openh323.org/ - cannot compile
  
   Greetings.
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Your suggestions helped.
Thanks a lot!
Greetings
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[Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls

2006-06-21 Thread Leah Newmark

Just saw this message back to me...I didn't realize my message was
posted and I ended up posting again. Oops!

Anyway, she *is* able to receive calls. She gets a fast busy when trying
to dial anything.

I know we had her do speed tests on her DSL the end of last year but I
don't remember the outcome, but that was for quality issues, so I don't
think it has to do with this problem per se.

Any ideas of what to test or look at?

Thanks!

LN

Message: 18
Date: Tue, 20 Jun 2006 00:12:51 +0200
From: lenz [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] User Loses Ability to Make Outgoing
   Calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; format=flowed; delsp=yes;
   charset=iso-8859-15


Hello Leah,
it may be the quality of her link degrading - it happens easily with ADSL.  
which error does she get? and she cannot receive calls at the same time,  
right?
l.


  

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RE: [Asterisk-Users] syntax error

 From: Rob Thomas
 
 That's freePBX or AMP code that we've since fixed - The 
 replacement line is
 
 exten = 
 s,2,GotoIf($[${CALLERID(name):0:${LEN(${RGPREFIX})}} != 
 ${RGPREFIX}]?4:3)  ; check for old prefix
Yes, ok. I'm gradually fixing all the code using Asterisk 1.2 syntax.

 (Upgrade to freePBX 2.1.1, it's much better, really)
I upgraded to custom, 'made with vi' files, thanks! This was an AMP
residue.

DV

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Re: [Asterisk-Users] TDM400P bad echo problem, tried lots of things

2006-06-21 Thread Andrei (MPI)


Try these settings in zapata.conf:

echocancel=64
echotraining=800
echocancelwhenbridged=yes
rxgain=3.2
txgain=-3.2

with default KB1 echo canceller in zconfig.h

This setup was working fairly okay for me for about a year or so.

Also, notice that at the first seconds of the call you may hear some 
echo, but then it disappears. I just trained myself to ignore this first 
seconds echo. =)


Please give the list information regarding the phone that you use with 
*. This maybe a cheap phone problem, as well.


Andrei (MPI)

Carey O'Shea wrote:

I have a bad echo problem on my TDM400P with one FXO module installed.

I have tried a few things, such as:

* setting rxgain and txgain to 0
* setting echocancelwhenbridged to no / yes
* settting echocancel to 64 / no / yes
* setting echocanceltraining to 800 / no / yes
* MG2 echo cancellation
* MARK2 echo cancellation
* KB1 echo cancellation
* AGGRESSIVE_SUPPRESSOR option of MARK2

Each time restarting Asterisk, then opening the Zap channel, and then
speaking...only to hear my self played back almost instantly. 


None of these options changed the echo for me, it always sounded the
same -- except for the AGGRESSIVE_SUPPRESSOR option, in which every time
I spoke it made the other end a very low volume, so much that I couldn't
hear the other end (ie: not useful).

I don't have this problem with pure IP calls, it's only with my TDM400P
and FXO that I have this echo problem. This means my headset and IP
phones are fine (of course).

So, what else can I try? :-)

Any ideas why this is so consistent and persistent? Maybe it's something
to do with my phone cable or something of that nature (hmm?)?

Any input appreciated.

Thanks,
Carey O'Shea.


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RE: [Asterisk-Users] zapata.conf: recent changes?

 From: Rob Thomas
 
 Looks like you've stopped compiling libpri. All those options 
 that are being ignored, are being ignored because they're for 
 PRI, and you don't have PRI support in zaptel.
Uh? 
If I don't have PRI support in zaptel, how are my 80 employees calling their
homes now?!

:-)

DV

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Re: [Asterisk-Users] forward a call to a SIP account on a remote server

2006-06-21 Thread William Piper

Close but not quite. Try below:

1. Setup sip.conf in theremote server to direct the call to the correct context
 [incoming]
host=(xxx.yyy.zzz.xxx)IP of the sending servertype=friend context=(context that is holding theexten for the user) allow=ulaw

2.Setup extensions.conf on theremote serverlike so:
 exten = 33347563,1,Dial,SIP/user

3. Setup extensions.conf on the forwarding server like so: 
 exten = 3347563,1,Dial,SIP/[EMAIL PROTECTED]
bp
On 6/21/06, nik600 [EMAIL PROTECTED] wrote:
Hii will forward a call to a remote server (only for one account)is this sintax correct?
exten = 33347563,1,Dial(SIP/[EMAIL PROTECTED])thanks___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] zapata.conf: recent changes?

2006-06-21 Thread Michiel van Baak

On 15:12, Wed 21 Jun 06, Tzafrir Cohen wrote:
 On Wed, Jun 21, 2006 at 01:57:36PM +0200, Mimmus wrote:
  Hi,
  after a few of upgrades, I noticed these messages in full debug log:
  
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring internationalprefix
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring nationalprefix
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring localprefix
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring privateprefix
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring unknownprefix
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring priindication
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring facilityenable
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling
  Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling
 
 Missing '[channels]' ?
 
 Could you provide the complete file?

Did you see this after a reload?
Asterisk will ignore some settings when doing a reload.
Only a restart will pickup changes to the settings mentioned in your mail.

-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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[Asterisk-Users] SIP or IAX client written in C

2006-06-21 Thread asterisk


I am looking to make a linux application that will use a SIP or IAX
clinet to connect to my Asterisk server and make calls.

I would like it to be written in C, but beggers can't be choosers. Any
information that would help me with my development would be appreciated.

If you know of a project that is already basically made, feel free to
point me in the right direction and I will be off your back.

Regards,

Chris Andrist

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[Asterisk-Users] AMD Machine Detect

2006-06-21 Thread Al Lougher

Hi -I have been developing an auto-dialling application similar to Voiceshot, Call-em-all etc. The one thing I am now struggling to get working is the ability to leave a message on an answering machine or cell phone voicemail. I am using app_amd.c and while it works well for some phones it is proving to be very difficult tweaking the settings to get it to work reliable enough to go to production. If anyone is using this successfully in a production environment I would really appreciate any posts of settings you are using. My settings are as follows:AMD(3500|1500|300|5000|120|50|5|256)Thank you.  Alan. 
		Want to be your own boss? Learn how on  Yahoo! Small Business. 
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Re: [Asterisk-Users] Anyone using VoIP WiFi phones?

2006-06-21 Thread Cory Andrews




Yeah customer feedback on these types of devices 
ranges from love to hate, it boils down to individual user preference. 
>From what I have seen, the Skype API is pretty tightly built into the 
Communicator, I am not certain if it will be able to be used with any softphone, 
that remains to be seen.

ClearOne has a nice USB speakerphone as well, the 
Chat50. I use it every day with Skype, Gizmo, etc, works well with every 
softphone interface I have tried. No BlueTooth, but very nice audio 
quality.

I am also testing a new product from Siemens called 
the C450 IP, which is dual mode PSTN/VoIP, and allows you to toggle between your 
PSTN line and VoIP. Has a base station and a wireless DECT handset, 
similar to the Aastra 480i-CT, but scaled down a bit in price and feature 
set.

Cory J AndrewsVOIPSupply.com454 Sonwil 
DriveBuffalo, NY 14225++voice - 716.630.1555 
X22email - [EMAIL PROTECTED]AIM - B2CORY

  - Original Message - 
  From: 
  Michael Graves 
  
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, June 21, 2006 11:46 
  AM
  Subject: Re: [Asterisk-Users] Anyone 
  using VoIP WiFi phones?
  Sorry to be 
  contrarian..but I bought the MV900 (from you actually) and I'm not really 
  impressed with it. Prior to it I had been using the Phoenix Audio Duet which 
  is a vastly superior device, albeit lacking the bluetooth 
  capability.As soon as the new Polycom Communicator is available I 
  expect I'll try that and pawn the MV900 off on some unsuspecting 
  colleague.MichaelOn Wed, 21 Jun 2006 10:25:41 -0400, Cory 
  Andrews wrote:MVox makes a BlueTooth enabled speakerphone that 
  works nicely with softphones, but is not inexpensive. Details can be 
  found here http://www.mvox.com/Cory J 
  AndrewsVOIPSupply.com454 Sonwil 
  DriveBuffalo, NY 14225++voice - 
  716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY- Original Message - 
  From: "Andrew Kohlsmith" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Wednesday, June 21, 2006 9:38 
  AMSubject: Re: [Asterisk-Users] Anyone using VoIP WiFi 
  phones? On Tuesday 20 June 2006 17:28, Karl J. 
  Vesterling wrote: I could care less about the color LCD, or 
  browsing the web on the phone, but a good speakerphone built 
  into a cordless WiFi is definitely a requirement that needs to 
  be met before I purchase one. Agreed, although every 
  single manufacturer is missing out on Bluetooth.  This 
  is a requirement for us. Corded headsets are such a pain in the 
  ass, especially on cordless phones. 
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Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing Calls

2006-06-21 Thread Michael Welter

Look in the sip.conf (or whatever) and make sure the context specifies 
a context that allows outgoing calls.


Leah Newmark wrote:

Just saw this message back to me...I didn't realize my message was
posted and I ended up posting again. Oops!

Anyway, she *is* able to receive calls. She gets a fast busy when trying
to dial anything.

I know we had her do speed tests on her DSL the end of last year but I
don't remember the outcome, but that was for quality issues, so I don't
think it has to do with this problem per se.

Any ideas of what to test or look at?

Thanks!

LN


Message: 18
Date: Tue, 20 Jun 2006 00:12:51 +0200
From: lenz [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] User Loses Ability to Make Outgoing
Calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; format=flowed; delsp=yes;
charset=iso-8859-15


Hello Leah,
it may be the quality of her link degrading - it happens easily with ADSL.  
which error does she get? and she cannot receive calls at the same time,  
right?

l.


 


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--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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Re: [Asterisk-Users] USB handset options for softphones

2006-06-21 Thread Warren





I am assuming you are talking about the Skype handsets here... Which
soft phones do these work with? Any linux ones?

W

Michael Graves wrote:
Voip Supply
has a number of USB handsets available... see
http://www.voipsupply.com/index.php?cPath=95_258 I'm not affiliated
with them beyond being a happy customer.
  
There are also a few USB handsets that look like desk phones. I have
seen these offered in the $50-70 range, some even with speaker phones.
Not sure where though. Try a quick search on Ebay.
  
Michael
  
  
On Wed, 21 Jun 2006 08:19:44 -0400, klubarpop wrote:
  
Are there any traditional telephone set-looking handset for use
with a
softphone? All the options I've found are the headset type. I'm
looking
for something more traditional--it should look like a small deskset,
cellphone or cordless phone, perhaps with a dial pad and a couple
of buttons
that integrates well with a softphone client. My users don't want
to pick
up a headset/mic when the phone rings or use the built-in computer
speaker/mic. 

I've looked at the Clarisys i750H, but it's not yet shipping. Any
other
popular options? And softphone clients do you use with them?

Many thanks
Ken lubar

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Re: [Asterisk-Users] database copy in asterisk

2006-06-21 Thread Shenen Shenen

Yes I want only copy the asterisk database, and I want that the authentificated sip registrations, work on the second asteriskAtHome like in the first,if I make this;my problem is:I want use softphones or wi_fi cell in 2 different asteriskAtHome, becouse I can't register sip on more asteriskAtHome, becouse I have 1 only proxy to set in the softphones or in some wi_fi (only with X-lite or Pro I can set more proxies...)so I want that the sips work on all 2 asteriskAtHome (for the priority of the 2 isn't a problem becouse I use the vrrpd), if I closed one pc I want that the other one is still active and I can call from the other one.
Could I copy all the asterisk database to one asteriskAtHome to the other or there are some files that are in conflict?can I do this?On 6/21/06, Time Bandit
 [EMAIL PROTECTED] wrote:
 Which files must I copy?then..I'll use a ssh scritp for this, I want only know which files I must copy...the MySQL files are usually in /var/lib/mysql. The databse you wantto copy is asterisk
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[Asterisk-Users] asterisk compiling

2006-06-21 Thread Giordano Grandis




Anyone can help with 
this?

cli.c:49:30: 
asterisk/version.h: No such file or directorycli.c: In function 
`handle_version':cli.c:414: error: `ASTERISK_VERSION' undeclared (first use 
in this function)cli.c:414: error: (Each undeclared identifier is reported 
only oncecli.c:414: error: for each function it appears in.)make: *** 
[cli.o] Error 
1ASTERISK 
installed. 
Installation 
finished.
I'm compiling 
bristuff-0.3.0-PRE-1q (asterisk-1.2.9.1)
Libpri, zaptel 
and zaphfc compiled witout problem.

Thanks


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

2006-06-21 Thread Martin Joseph



On Jun 21, 2006, at 5:19 AM, klubarpop wrote:


Are there any traditional telephone set-looking handset for use with a
softphone?  All the options I've found are the headset type.  I'm 
looking

for something more traditional--it should look like a small deskset,
cellphone or cordless phone, perhaps with a dial pad and a couple of 
buttons
that integrates well with a softphone client.  My users don't want to 
pick

up a headset/mic when the phone rings or use the built-in computer
speaker/mic.

I think figuring out which softphone is going to work for you is most 
important.  If you expect to be able to dial from a usb phone, the 
softphone needs to specifically support the keypad.


I have experimented with Diax, which is a very tiny windows app(I like 
it lots), and the yealink USB-P1K.  This handset is supposedly 
supported by Diax, but it has some issues including an issue where the 
user needs to use the computer to answer calls (the green button on the 
handset doesn't work).  This is kind of annoying, but hopefully soon to 
be fixed.


The sound quality of this phone is pretty amazing in my opinion, as 
it's audio is cleaner then any other source I have heard (not that 
many).  I suspect this due to some audio processing (noise suppression) 
in the handset proper.


Anyhow Diax is very promising, and I am hoping for a timely update that 
helps my problems with the USB handset I have purchased...


Marty



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Re: [Asterisk-Users] TE420P/TE415P?

2006-06-21 Thread C F


I like the TC400P card, how many T1s will that take? or is it just a
Daughter card on the TE4xx ? How many channels can it transcode?

On 6/20/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

Hi,

I just read a pressrelease from VON that Digium will soon be releaseing
a couple of new cards. What got me interested was: The TE420P and
TE415P support 128ms of G.168 (2002)-compliant echo cancellation across
their entire 128 channels.

Does anyone know when thease will be released and what they will cost
when released? Thanks!

http://pressroom.pulvermedia.com/digium/pr.php#0314c

Regards,
jan
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[Asterisk-Users] Re: Avaya phone 4610sw message waiting indicator and other settings

2006-06-21 Thread Erick Perez


nobody uses avaya phones with asterisk?

On 6/20/06, Erick Perez [EMAIL PROTECTED] wrote:

Hi, I setup my tftp to send SIP configurations (the bin files) to the
avaya phone. When it finished loading and rebooting it asked for the
extension and the password and the asterisk ip address. I had to input
that manually and is now working perfectly with asterisk.

what is the format of the text files to make this phone load the
asterisk ip, extension number, codec used, password as well as to
configure message waiting indicator and maybe modify some of the
buttons (such as just pressing one of the available programmable
buttons to access voicemail). I have 10 more of these phones and i
want to do provisioning automatically.

in the 46xxsettings.txt file there are no such parameters


thanks,


--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780





--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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Re: [Asterisk-Users] SIP or IAX client written in C

2006-06-21 Thread Time Bandit


I am looking to make a linux application that will use a SIP or IAX
clinet to connect to my Asterisk server and make calls.

I would like it to be written in C, but beggers can't be choosers. Any
information that would help me with my development would be appreciated.

If you know of a project that is already basically made, feel free to
point me in the right direction and I will be off your back.

start here : http://iaxclient.sourceforge.net/
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Re: [Asterisk-Users] Asterisk queue log solution?


Does a solution exist that I am overlooking that may provide the
functionality I am after?


I don't understand why Queue-Metrics will not do what you need?   We
run it and it does everything you just said you wanted to do.
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[Asterisk-Users] Compiling asterisk

2006-06-21 Thread Giordano Grandis





Anyone can help with 
this?

cli.c:49:30: 
asterisk/version.h: No such file or directorycli.c: In function 
`handle_version':cli.c:414: error: `ASTERISK_VERSION' undeclared (first use 
in this function)cli.c:414: error: (Each undeclared identifier is reported 
only oncecli.c:414: error: for each function it appears in.)make: *** 
[cli.o] Error 
1ASTERISK 
installed. 
Installation 
finished.
I'm compiling 
bristuff-0.3.0-PRE-1q (asterisk-1.2.9.1)
Libpri, zaptel 
and zaphfc compiled witout problem.

Thanks


Giordano




  
  

  


  
Giordano 
Grandise-mail : [EMAIL PROTECTED]VoIP: 
sip:[EMAIL PROTECTED]
  


  
_www.invidea.it
  TecnoJest SrlVerrotti c/o Centro 
  attività "Espansione II, int 4", 65016 Montesilvano (PE)Tel [+39] 085 4450011- Fax [+39] 
  085 4459477 - PI 01635460684

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[Asterisk-Users] Polycom 601 problems with multiple registrations

2006-06-21 Thread Brian Vincent $C$









Im stumped on this one and any help would be greatly
appreciated.



Im just trying to get my Polycom 601 to have multiple
extensions on it. For example, on line 1 I want extension 21, on line 2 I
want extension 22, and on line 3 I want extension 23. Ideally Id
actually have each extension appear on 2 lines and therefore filling up all
6. I should be able to do that with the reg.x.lineKeys parameter.
Anyway, Im not even at the point of getting multiple registrations to
work, so Ill worry about that later. Right now the only thing that
works is registering the first extension  it registers just fine and
works as expected. No matter what extension I put on there it works, but
I only have line 1 working. What am I doing wrong?



Okay, now my config. Ive got a REALLY basic set
up. I copied the files off the wiki from krisk.org. I completely
removed ipmid.cfg temporarily so it wouldnt interfere with this (putting
it back in place has no effect). That leaves me with just sip.cfg and the
phone cfg file. Im booting with FTP. I know the config files
are loading correctly because I can make changes and they do have an
effect. Heres the phone20.cfg file for the phone:



?xml version=1.0 encoding=UTF-8
standalone=yes?

!-- Example Per-phone Configuration File --

!-- $Revision: 1.59 $ $Date: 2004/05/22 00:50:41 $ --

phone1

reg

 reg.1.address=21

 reg.1.auth.userId=21

 reg.1.auth.password=21

 reg.1.server.1.address=10.20.0.1

 reg.2.address=22

 reg.2.auth.userId=22

 reg.2.auth.password=22

 reg.2.server.1.address=10.20.0.1

 reg.3.address=23

 reg.3.auth.userId=23

 reg.3.auth.password=23

 reg.3.server.1.address=10.20.0.1 /



/phone1



And sip.cfg:



!-- IP Application Configuration File --

!--

$Revision: 1.57.2.1 $ $Date: 2004/07/27 00:23:34 $

--



sip



voIpProt

local voIpProt.local.port=5060/

server voIpProt.server.1.address=10.20.0.1
voIpProt.server.1.port=5060 voIp

Prot.server.1.transport=UDPonly
voIpProt.server.1.expires=3600 voIpProt.serv

er.1.register=1
voIpProt.server.1.retryTimeOut=0 voIpProt.server.1.retryMaxC

ount=0
voIpProt.server.1.expires.lineSeize=30/



 SIP
voIpProt.SIP.useRFC2543hold=1 voIpProt.SIP.lcs=0 voIpProt.SIP.sendCompactHdrs=0
voIpProt.SIP.WM50=0 voIpProt.SIP.keepalive.sessionTimers=0
voIpProt.SIP.requestURI.E164.addGlobalPrefix=

outboundProxy voIpProt.SIP.outboundProxy.address=
voIpProt.SIP.outboundProxy.port=5060/

alertInfo voIpProt.SIP.alertInfo.1.value=AA
voIpProt.SIP.alertInfo.1.class=3 /

alertInfo voIpProt.SIP.alertInfo.2.value=RA
voIpProt.SIP.alertInfo.2.class=4 /



 requestValidation
voIpProt.SIP.requestValidation.1.request= voIpProt.SIP.requestValidation.1.method=
voIpProt.SIP.requestValidation.1.request.1.event=

digest voIpProt.SIP.requestValidation.digest.realm=10.20.0.1/

/requestValidation

specialEvent voIpProt.SIP.specialEvent.lineSeize.nonStandard=1
voIpProt.SIP.specialEvent.checkSync.alwaysReboot=0/

conference voIpProt.SIP.conference.address=/

/SIP

/voIpProt



 dialplan dialplan.impossibleMatchHandling=2
dialplan.removeEndOfDial=

1

digitmap dialplan.digitmap=[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]xx

xxx|[2-9]xxxT dialplan.digitmap.timeOut=3/



 routing

server dialplan.routing.server.1.address=
dialplan.routing.server.1.port=506

0/

emergency
dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1

.server.1=1/

/routing

/dialplan



 logging



 level

change log.level.change.sip=4 log.level.change.sip.obs=5/

/level

/logging

/sip

---
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED] 





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RE: [Asterisk-Users] zapata.conf: recent changes?


 Did you see this after a reload?
 Asterisk will ignore some settings when doing a reload.
 Only a restart will pickup changes to the settings mentioned 
 in your mail.
True. In fact, after a restart, I don't see any WARNING.

Thanks
DV

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[Asterisk-Users] Re: faxdetect questions - Please HELP!

2006-06-21 Thread M.Hockings

In reading this problem and the solution I wonder if it would be better, 
(or acceptable to the users?) to send the received Hylafax fax to a 
(network attached) printer of their choice.  That way you eliminate the 
need for special equipment (the fax machine) ?


I did a bit of research and found :
http://www.hylafax.org/content/Handbook:Server_Operation:Tweaking_and_Customization
That has a section on Automatic Fax Printing that actually address 
this sort or thing.


Mike

Craig Guy wrote:

Hi Bob,

in order to stop fax detection, send the call to a context without a 
'fax' extension:


[incoming]

_.,1,doSomeStuff

; Hardfax extension
12345678,1,Goto(hardfax,1000,1)

fax,1,receiveFax

[hardfax]
1000,1,Dial(Zap/1|70)
1000,n,Hangup

- Original Message - From: Bob McDowell 
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Tuesday, June 20, 2006 6:23 AM
Subject: [Asterisk-Users] faxdetect questions - Please HELP!


I'm using IAXmodem and Hylafax with 'faxdetect=incoming' and things 
mostly
work pretty well.  My main lines come in via T1 DID.  Today, HR got 
tired of
having someone read and forward their faxes to them and requested we 
bring

their physical machine back on line.  I have been able to get the fax
forwarded to the appropriate zap channel, but I cannot seem to get it to
stop 'faxdetect'ing.  After deciding that it is a fax and sending it 
to the

proper zap channel Asterisk says:

   -- Executing Dial(Zap/5-1, Zap/105) in new stack
   -- Called 105
   -- Zap/105-1 is ringing
   -- Redirecting Zap/5-1 to fax extension
   -- Hungup 'Zap/105-1'

...and Hylafax gets it...

Now the questions:

1) How can I have 'faxdetect=incoming' for my T1 context and 
'faxdetect=no'

for my internal zap channels.  (I'm assuming that this is what's wrong
here...)
2) Is it instead possible to disable faxdetect for the duration of the
call?  E.g. exten = fax,1,zapFAXDETECT(off)
3) Is there a better way to mix detected faxes and dedicated fax lines?
4) Can anyone share with me a config that accomplishes this feat (both
detected and dedicated)?




 


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Re: [Asterisk-Users] Unicall acting really funny

2006-06-21 Thread Joao Mesquita


Hello Steve,

   Thank you for being so active helping people with Unicall problems,
I am sure a lot of us appreciate this.

   I could tweak a little bit more with your software versions from the
download site and I got half of the problems solved. Now I am able to
set the loglevel to 255 and that gives me a whole lot more tools to work
with. I am now also able to reload the chan_unicall.so module correctly
without having to restart asterisk. I am sorry for the lame mistakes.

   I am pretty sure that the CRC4 was not the problem because the FAS
increased with CRC4 on and I got a RED signal on the board. Logs on that
are printed below.

   That leaves me with 2 problems and 1 warning.

   The first problem is that some phones on the office (the Siemens
Euroset family) are not able to complete calls from the Highpath to the
asterisk (dialing to a SIP extension). I got a T3 timed out message and
that's where the call hangs. I could get to this conclusion because all
other phone models I tested with completed the calls just fine. I don't
know how many other phone models on the office present the same problem.
Logs are below.

   The second problem is when I try to pickup a call coming from the
Highpath and dial the extension number dialed to the PSTN E1. The call
does not complete although it gets to ring on the destination (on this
case my mobile phone) once. The most funky part of this is that this
happens MOST of the times, but not all of them and this is what was
suggesting a CRC4 problem (I guess). The verbose version of the log
files are below.

   The last is a warning might be related to the second problem. Every
time I make a call from a SIP endpoint to the PSTN using my PSTN E1 and
the PSTN phone disconnects the call, I get an cause 32773 - Unexpected
CAS bit pattern message. I call this a warning because I could not
relate any real problem on the calls with that. Logs on that are also below.

   Thank you in advance,

   Mesquita

* CRC4 / RED SIGNAL LOGS ***

asterisk-test:~# cat /proc/zaptel/1
Span 1: Tor2/0/1 Tormenta 2 (PCI) Quad E1 Card 0 Span 1 HDB3//CRC4 RED
   CRC4 error count: 986
   FAS error count: 192

  1 Tor2/0/1/1 CAS

asterisk-test:~# cat /proc/zaptel/2
Span 2: Tor2/0/2 Tormenta 2 (PCI) Quad E1 Card 0 Span 2 HDB3//CRC4

 32 Tor2/0/2/1 CAS
 33 Tor2/0/2/2 CAS


   -- Executing Dial(SIP/teste-4640, Unicall/g1/94109570) in new stack
Jun 13 03:36:35 WARNING[1874]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Call control(1)
Jun 13 03:36:35 WARNING[1874]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Make call
Jun 13 03:36:35 WARNING[1874]: chan_unicall.c:1077 unicall_call: Make
call failed - Blocked
   -- Couldn't call g1/94109570
Jun 13 03:36:35 WARNING[1874]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Channel gains
Jun 13 03:36:35 WARNING[1874]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Channel switching
   -- Hungup 'UniCall/1-1'
 == Everyone is busy/congested at this time (0:0/0/0)



** FIRST PROBLEM LOGS *

asterisk-test:~# cat /proc/zaptel/1
Span 1: Tor2/0/1 Tormenta 2 (PCI) Quad E1 Card 0 Span 1 HDB3/ ClockSource

  1 Tor2/0/1/1 CAS (In use)
  2 Tor2/0/1/2 CAS (In use)

asterisk-test:~# cat /etc/asterisk/extensions.conf (snip)
[from-e1-interno]
exten = _X.,1,Answer
exten = _X.,n,Dial(SIP/teste)
exten = _X.,n,Hangup

Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39  - 0001  [1/   1/Idle  /Idle ]
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39 Detected
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39 Making a new call with CRN 32769
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39 1101  -  [2/   2/Idle  /Idle ]
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:2644 handle_uc_event:
Unicall/39 event Detected
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39  - 1 on  [2/   2/Seize ack /Seize ack]
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39 5 on  -  [2/   2/Seize ack /Seize ack]
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39  - 1 off [2/   2/Group A   /Category req ]
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39 5 off -  [2/   2/Group A   /Category req ]
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39  - 1 on  [2/   2/Group A   /Category req ]
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39 5 on  -  [2/   2/Group A   /Category req ]
Jun 13 03:24:03 WARNING[1686]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/39  - 1 off [2/   2/Group A   /ANI request  ]
Jun 13

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

2006-06-21 Thread Cory Andrews

Not sure why you were told the Clarisys i750H is
not shipping, we've been working with these for quite some time, and have seen
good availability on them.

Thanks

Cory J AndrewsVOIPSupply.com454 Sonwil
DriveBuffalo, NY 14225++voice - 716.630.1555
X22email - [EMAIL PROTECTED]AIM - B2CORY

- Original Message -
From:
Warren
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, June 21, 2006 12:30
PM
Subject: Re: [Asterisk-Users] USB handset
options for softphones
I am assuming you are talking about the Skype handsets
here... Which soft phones do these work with? Any linux
ones?WMichael Graves wrote:
Voip Supply has a
number of USB handsets available... see http://www.voipsupply.com/index.php?cPath=95_258
I'm not affiliated with them beyond being a happy customer.There are
also a few USB handsets that look like desk phones. I have seen these
offered in the $50-70 range, some even with speaker phones. Not sure where
though. Try a quick search on Ebay.MichaelOn Wed, 21 Jun
2006 08:19:44 -0400, klubarpop wrote:Are there any traditional
telephone set-looking handset for use with asoftphone? All the
options I've found are the headset type. I'm lookingfor something
more traditional--it should look like a small deskset,cellphone or
cordless phone, perhaps with a dial pad and a couple of buttonsthat
integrates well with a softphone client. My users don't want to
pickup a headset/mic when the phone rings or use the built-in
computerspeaker/mic. I've looked at the Clarisys
i750H, but it's not yet shipping. Any otherpopular options? And
softphone clients do you use with them?Many
thanksKen
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Re: [Asterisk-Users] USB handset options for softphones