date:20070228



I have a client intersted in a system, but they have an ISDN30 line - the 
down side is that I've not done any before...


Now, I've no reason to think it won't work, but as going on-site with a 
new card and no first-hand knowledge isn't particularly wise, I'm 
wondering about the best way to do some testing...


So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30 willing 
to let me play? ;-)


Failing that, what test gear exists to pretend to be an exchange line? 
(Although I'm suspecting it's going to be outside my budget )-:


But finally, can you run a TE110P card in master mode? ie. can I get 2 
of these, and put one in a separate box and use it to pretend to be the 
(BT) exchange, with the other box doing what it's supposed to do?


If it is possible, what do I need in the way of cross-over cables, etc.?

Thanks,

Gordon
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[asterisk-users] groups

2007-02-28 Thread Khaled

Dears 

 

Please how can create an independent group of users on asterisk ,in which
user on group A cant dial user on group B.

 

 

Thanks 

 

 

 

 

 

Khaled Chehab

System Integration Engineer

Xplorium Offshore.

Sakiet Al Janzir

Postal Code: 1102-2080

Tel: (961) 1- 868 686

Fax :(961) 1-808 810

GSM: (961) 3-979 343

 




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Re: [asterisk-users] Re: queue information into db

2007-02-28 Thread nik600


On 2/28/07, Tomislav Parcina [EMAIL PROTECTED] wrote:

nik600 wrote:
 In the last months i've developed a web application for the use of an
 asterisk call center.

 Yuo can
 - make calls from a web interface
 - login/logout in queue
 - view members logged in a queue
 - view callers queued in a queue
 - pickup a callers from a queue

What is license of this application? Can it be downloaded from somewhere?


--
Tomislav Parcina
[EMAIL PROTECTED]

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actually it isnìt released under any type of licence.
if you want i can put the code on my web site
(but no earlier than the next week)
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Re: [asterisk-users] ISDN30 testing questions ...

2007-02-28 Thread tim robinson


Hi Gordon

'Fraid I don't have a line you can 'play with' so to speak! However, 
firstly, I have installed several E1 based Asterisk systems both in UK 
and elsewhere, and apart from a few telco issues in a Latin American 
country, it just works.


You can do as you suggest, i.e. have two Asterisk boxes back to back, 
and we have done this on many occasions for load-testing our systems. To 
do so you will need to make an E1 crossover with pins 1/2 going to pins 
4/5 at the other end.


Good luck

Tim Robinson
Basingstoke, UK



Gordon Henderson wrote:


I have a client intersted in a system, but they have an ISDN30 line - 
the down side is that I've not done any before...


Now, I've no reason to think it won't work, but as going on-site with 
a new card and no first-hand knowledge isn't particularly wise, I'm 
wondering about the best way to do some testing...


So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30 
willing to let me play? ;-)


Failing that, what test gear exists to pretend to be an exchange line? 
(Although I'm suspecting it's going to be outside my budget )-:


But finally, can you run a TE110P card in master mode? ie. can I get 
2 of these, and put one in a separate box and use it to pretend to be 
the (BT) exchange, with the other box doing what it's supposed to do?


If it is possible, what do I need in the way of cross-over cables, etc.?

Thanks,

Gordon
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Re: [asterisk-users] groups

2007-02-28 Thread Mohamed A. Gombolaty

Dear Khaled,

The way I would go to do so is to put  the group of people you want to
call each other in one context and the other people in an another
context. That's one way to do so.

Thx
MAG

Khaled wrote:

 Dears

 Please how can create an independent group of users on asterisk ,in
 which user on group A cant dial user on group B.

 Thanks

 Khaled Chehab

 System Integration Engineer

 Xplorium Offshore.

 Sakiet Al Janzir

 Postal Code: 1102-2080

 Tel: (961) 1- 868 686

 Fax :(961) 1-808 810

 GSM: (961) 3-979 343

 ---
 *
 No employee or agent is authorized to conclude any binding agreement
 on behalf of Xplorium with another party by e-mail without express
 written confirmation by an officer of Xplorium. Any views expressed by
 an individual in this electronic message do not necessarily reflect
 views of Xplorium or its subsidiaries and associates.

 This electronic message and its attachments are solely addressed to
 the addressee(s), and contain confidential information protected from
 disclosure belonging to Xplorium.

 If you are not the intended addressee of this electronic message and
 its attachments, kindly delete it immediately from your system and
 notify the sender by electronic mail. You must not copy this message
 or attachment or disclose its content to any other person.

 Xplorium does not guarantee the integrity of this electronic message
 and any of its attachments, or that they are free from computer
 viruses or other defects.
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--
Thx
MAG


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Re: [asterisk-users] ISDN30 testing questions ...

Hmm, I am in England too on the East London corner. Tell me what you are about 
to do with the
ISDN30 in relation with your TE110p? It is not clear how you would set this up 
based on your
e-mail. Be specific, I might be able to help you. Explain more how your client 
wished to have you
work on.


 I have a client intersted in a system, but they have an ISDN30 line - the
 down side is that I've not done any before...

 Now, I've no reason to think it won't work, but as going on-site with a
 new card and no first-hand knowledge isn't particularly wise, I'm
 wondering about the best way to do some testing...

 So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30 willing
 to let me play? ;-)

 Failing that, what test gear exists to pretend to be an exchange line?
 (Although I'm suspecting it's going to be outside my budget )-:

 But finally, can you run a TE110P card in master mode? ie. can I get 2
 of these, and put one in a separate box and use it to pretend to be the
 (BT) exchange, with the other box doing what it's supposed to do?

 If it is possible, what do I need in the way of cross-over cables, etc.?

 Thanks,

 Gordon
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Re: [asterisk-users] ISDN30 testing questions ...

Hi,

What is the main purpose of this setup by the way?

 Hi Gordon

 'Fraid I don't have a line you can 'play with' so to speak! However,
 firstly, I have installed several E1 based Asterisk systems both in UK
 and elsewhere, and apart from a few telco issues in a Latin American
 country, it just works.

 You can do as you suggest, i.e. have two Asterisk boxes back to back,
 and we have done this on many occasions for load-testing our systems. To
 do so you will need to make an E1 crossover with pins 1/2 going to pins
 4/5 at the other end.

 Good luck

 Tim Robinson
 Basingstoke, UK



 Gordon Henderson wrote:

 I have a client intersted in a system, but they have an ISDN30 line -
 the down side is that I've not done any before...

 Now, I've no reason to think it won't work, but as going on-site with
 a new card and no first-hand knowledge isn't particularly wise, I'm
 wondering about the best way to do some testing...

 So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30
 willing to let me play? ;-)

 Failing that, what test gear exists to pretend to be an exchange line?
 (Although I'm suspecting it's going to be outside my budget )-:

 But finally, can you run a TE110P card in master mode? ie. can I get
 2 of these, and put one in a separate box and use it to pretend to be
 the (BT) exchange, with the other box doing what it's supposed to do?

 If it is possible, what do I need in the way of cross-over cables, etc.?

 Thanks,

 Gordon
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Re: [asterisk-users] ISDN30 testing questions ...


On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:


Hi,

What is the main purpose of this setup by the way?


For me? To provde a client with a VoIP capable PBX in their office to 
replace their current steam driven PBX...


(And hopefully to earn a few Â£Â£in the process!)

I have a lot of installations now with pure analogue lines, now want to do 
one with a digital line, but I want to test it out before going on-site, 
so I don't look like an idiot when it doesn't work...


Gordon




Hi Gordon

'Fraid I don't have a line you can 'play with' so to speak! However,
firstly, I have installed several E1 based Asterisk systems both in UK
and elsewhere, and apart from a few telco issues in a Latin American
country, it just works.

You can do as you suggest, i.e. have two Asterisk boxes back to back,
and we have done this on many occasions for load-testing our systems. To
do so you will need to make an E1 crossover with pins 1/2 going to pins
4/5 at the other end.

Good luck

Tim Robinson
Basingstoke, UK



Gordon Henderson wrote:


I have a client intersted in a system, but they have an ISDN30 line -
the down side is that I've not done any before...

Now, I've no reason to think it won't work, but as going on-site with
a new card and no first-hand knowledge isn't particularly wise, I'm
wondering about the best way to do some testing...

So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30
willing to let me play? ;-)

Failing that, what test gear exists to pretend to be an exchange line?
(Although I'm suspecting it's going to be outside my budget )-:

But finally, can you run a TE110P card in master mode? ie. can I get
2 of these, and put one in a separate box and use it to pretend to be
the (BT) exchange, with the other box doing what it's supposed to do?

If it is possible, what do I need in the way of cross-over cables, etc.?

Thanks,

Gordon
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Re: [asterisk-users] Re: queue information into db

2007-02-28 Thread David Boyd

On Wed, 2007-02-28 at 13:05 +0100, nik600 wrote:
 On 2/28/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
  nik600 wrote:
   In the last months i've developed a web application for the use of an
   asterisk call center.
  
   Yuo can
   - make calls from a web interface
   - login/logout in queue
   - view members logged in a queue
   - view callers queued in a queue
   - pickup a callers from a queue
 
  What is license of this application? Can it be downloaded from somewhere?
 
 
  --
  Tomislav Parcina
  [EMAIL PROTECTED]
 
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 actually it isnìt released under any type of licence.
 if you want i can put the code on my web site
 (but no earlier than the next week)
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That would be great, can you provide a URL when it is available.  This
would greatly assist us in our trouble handing scenario.



db

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Re: [asterisk-users] ISDN30 testing questions ...


On Wed, 28 Feb 2007, tim robinson wrote:


Hi Gordon

'Fraid I don't have a line you can 'play with' so to speak! However, firstly, 
I have installed several E1 based Asterisk systems both in UK and elsewhere, 
and apart from a few telco issues in a Latin American country, it just works.


Thanks! I've no reason to think it wouldn't just work, but it's always 
nice to have had some first-hand experience beforehand...


You can do as you suggest, i.e. have two Asterisk boxes back to back, and we 
have done this on many occasions for load-testing our systems. To do so you 
will need to make an E1 crossover with pins 1/2 going to pins 4/5 at the 
other end.


Great. Thanks again.

Gordon
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Re: [asterisk-users] ISDN30 testing questions ...


On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:

Hmm, I am in England too on the East London corner. Tell me what you are 
about to do with the ISDN30 in relation with your TE110p? It is not 
clear how you would set this up based on your e-mail. Be specific, I 
might be able to help you. Explain more how your client wished to have 
you work on.


Have I selected the wrong card then? They have an incoming E1/ISDN30 line 
going into a prehistoric PBX which they want to replace with a shiny 
new VoIP capable PBX.


All I want to do is some tests beforehand, so in the abscence of a local 
site with an ISDN30 connection to play with, I want to connect 2 boxes 
back to back with a TE110p card in each box and a cross-over cable... One 
box will pretend to be the BT exchange, then other the CPE. On the BT 
exchange box, I will place a few calls, and have it route them over the 
line to the other box which will answer them (and vice versa)


Gordon







I have a client intersted in a system, but they have an ISDN30 line - the
down side is that I've not done any before...

Now, I've no reason to think it won't work, but as going on-site with a
new card and no first-hand knowledge isn't particularly wise, I'm
wondering about the best way to do some testing...

So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30 willing
to let me play? ;-)

Failing that, what test gear exists to pretend to be an exchange line?
(Although I'm suspecting it's going to be outside my budget )-:

But finally, can you run a TE110P card in master mode? ie. can I get 2
of these, and put one in a separate box and use it to pretend to be the
(BT) exchange, with the other box doing what it's supposed to do?

If it is possible, what do I need in the way of cross-over cables, etc.?

Thanks,

Gordon
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Re: [asterisk-users] running asterisk through cellphone

2007-02-28 Thread Dovid B

Cant take the credit. I didnt create it. as far as a phone you can go with 2 
things. either use chan_cellphone and use bluetooth or you can go with a cell 
phone dock (as some one mentioned earlier). if you are using the cellular 
docking station that you dont need to worry about chan_cellphone. in regards to 
your other question once the phone is set up it will act like a regular line. 
once asterisk is connected to it it's just a matter of setting up the dial plan 
to do what you want it to do.
  - Original Message - 
  From: Michael Kamleitner 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, February 27, 2007 9:21 PM
  Subject: Re: [asterisk-users] running asterisk through cellphone

  hi dovid,

  thx for replying, as I can see the chan_cellphone patch was done by you, 
great! looks like this is exactly what I want. my goal is to connect a normal 
consumer cellphone to the asterisk-server, allowing anyone else to phone-in 
from their regular phone. 

  it would be even better if I could use this setup to emulate extension - so 
lets assume my cellphone-number is 004369912345678, than I would like to have 3 
separate extensions at 004369912345678-01, 004369912345678-02 and 
004369912345678-03. is this possible? 

  as I'm going to buy a separate phone for this task, can anyone recommend 
certain models (besides the RIM blackberry mentioned in the docs)?

  greetings,
  michael

  On 2/27/07, Dovid B [EMAIL PROTECTED] wrote:
What is the cellular connection for ? Are you using this for inbound or the 
clients will call in in from thier cell phones ? If you need incoming (and or 
ourgoing) lines you can get one from an ITSP. If you want to use your cell 
phone you can use chan_cellphone. In order to use it you will need to install 
the patch. For more information have at look at this:
http://bugs.digium.com/view.php?id=8919 
http://bugs.digium.com/view.php?id=8919
  - Original Message - 
  From: Michael Kamleitner 
  To: asterisk-users@lists.digium.com 
  Sent: Tuesday, February 27, 2007 6:54 PM
  Subject: [asterisk-users] running asterisk through cellphone

  hi everybody,

  I'm currently planning a small-sized web-applicaiton allowing users to 
call-in via phone. the phonecalls should be recorded and processed further by 
some custom scripts - sounds like asterisk is a perfect match for this app. 

  however, during prototyping I have no ISDN-connection whatsoever 
available, so I was asking myself if it's possible to connect a cellphone via 
data-cable (or bluetooth?) and use this as the single line to call-in. 
searching the asterisk-forums I found mentions of chan_cellphone, which is 
probably a patch for exactly this kind of usage, right? 

  I'ld be thankful if you could just point me to the right direction (I'm 
quite new to asterisk). thx in advance! 

  michael

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  -- 
  10 Jahre The Gap
  Party am 15.3.2007 - www.tengap.at

  Mag. Michael Kamleitner
  - 
  [EMAIL PROTECTED]
  +43 699 11607923
  https://www.xing.com/profile/Michael_Kamleitner
  - 
  m-otion GmbH
  Favoritenstr 4-6/III, 1040 Wien
  +43 1 205705 / 21 (Fax 99)
  -
  www.m-otion.com 

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Re: [asterisk-users] ISDN30 testing questions ...

And how would you be able to make a test telephone call with ISDN30 when you 
don't have an E1
link? You gonna have two asterisk box connected peer to peer and have the other 
as the master and
the other as a slave. Or the other way of saying that your other asterisk box 
generates the
signalling/framing while the other is the recipient? How would you be able to 
know for sure if
this works?

 On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:

 Hi,

 What is the main purpose of this setup by the way?

 For me? To provde a client with a VoIP capable PBX in their office to
 replace their current steam driven PBX...

 (And hopefully to earn a few Â£Â£in the process!)

 I have a lot of installations now with pure analogue lines, now want to do
 one with a digital line, but I want to test it out before going on-site,
 so I don't look like an idiot when it doesn't work...

 Gordon


 Hi Gordon

 'Fraid I don't have a line you can 'play with' so to speak! However,
 firstly, I have installed several E1 based Asterisk systems both in UK
 and elsewhere, and apart from a few telco issues in a Latin American
 country, it just works.

 You can do as you suggest, i.e. have two Asterisk boxes back to back,
 and we have done this on many occasions for load-testing our systems. To
 do so you will need to make an E1 crossover with pins 1/2 going to pins
 4/5 at the other end.

 Good luck

 Tim Robinson
 Basingstoke, UK



 Gordon Henderson wrote:

 I have a client intersted in a system, but they have an ISDN30 line -
 the down side is that I've not done any before...

 Now, I've no reason to think it won't work, but as going on-site with
 a new card and no first-hand knowledge isn't particularly wise, I'm
 wondering about the best way to do some testing...

 So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30
 willing to let me play? ;-)

 Failing that, what test gear exists to pretend to be an exchange line?
 (Although I'm suspecting it's going to be outside my budget )-:

 But finally, can you run a TE110P card in master mode? ie. can I get
 2 of these, and put one in a separate box and use it to pretend to be
 the (BT) exchange, with the other box doing what it's supposed to do?

 If it is possible, what do I need in the way of cross-over cables, etc.?

 Thanks,

 Gordon
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Re: [asterisk-users] ISDN30 testing questions ...

This is perhaps an architectural issue. I suppose you are planning to interface 
the shining
asterisk-based VOIP box with their millenium old pabx? What is the brand 
name of their old PABX
machine though?

In my humble opinion, your setup to connect two asterisk box peer to peer using 
two TE110p wont
work in this scenario. Why? How you would be able to make a test phone calls?

 On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:

 Hmm, I am in England too on the East London corner. Tell me what you are
 about to do with the ISDN30 in relation with your TE110p? It is not
 clear how you would set this up based on your e-mail. Be specific, I
 might be able to help you. Explain more how your client wished to have
 you work on.

 Have I selected the wrong card then? They have an incoming E1/ISDN30 line
 going into a prehistoric PBX which they want to replace with a shiny
 new VoIP capable PBX.

 All I want to do is some tests beforehand, so in the abscence of a local
 site with an ISDN30 connection to play with, I want to connect 2 boxes
 back to back with a TE110p card in each box and a cross-over cable... One
 box will pretend to be the BT exchange, then other the CPE. On the BT
 exchange box, I will place a few calls, and have it route them over the
 line to the other box which will answer them (and vice versa)

 Gordon





 I have a client intersted in a system, but they have an ISDN30 line - the
 down side is that I've not done any before...

 Now, I've no reason to think it won't work, but as going on-site with a
 new card and no first-hand knowledge isn't particularly wise, I'm
 wondering about the best way to do some testing...

 So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30 willing
 to let me play? ;-)

 Failing that, what test gear exists to pretend to be an exchange line?
 (Although I'm suspecting it's going to be outside my budget )-:

 But finally, can you run a TE110P card in master mode? ie. can I get 2
 of these, and put one in a separate box and use it to pretend to be the
 (BT) exchange, with the other box doing what it's supposed to do?

 If it is possible, what do I need in the way of cross-over cables, etc.?

 Thanks,

 Gordon
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Re: [asterisk-users] ISDN30 testing questions ...


Gordon Henderson wrote:

On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:


Hi,

What is the main purpose of this setup by the way?


For me? To provde a client with a VoIP capable PBX in their office to 
replace their current steam driven PBX...


(And hopefully to earn a few Â£Â£in the process!)

I have a lot of installations now with pure analogue lines, now want 
to do one with a digital line, but I want to test it out before going 
on-site, so I don't look like an idiot when it doesn't work...


Gordon
You should be in for a treat then.  Digital is much easier IMO than 
analog setups. 

I have not touched an analog setup recently (about a year) but unless 
things have changed alot in echo cancellation and analog, that is what I 
spent most of my time doing, chasing intermittent echo.  I have yet to 
experience this with digital.  Not to say I have never encountered echo, 
I have, but after adjusted the echo is gone.


Also, there is a certain thrill to watch 30 channels come up one after 
another in rapid succession.


Thanks,
Steve Totaro
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Re: [asterisk-users] ISDN30 testing questions ...

Hi Tim,

What is the brand name of your existing PABX?

 Hi Gordon

 'Fraid I don't have a line you can 'play with' so to speak! However,
 firstly, I have installed several E1 based Asterisk systems both in UK
 and elsewhere, and apart from a few telco issues in a Latin American
 country, it just works.

 You can do as you suggest, i.e. have two Asterisk boxes back to back,
 and we have done this on many occasions for load-testing our systems. To
 do so you will need to make an E1 crossover with pins 1/2 going to pins
 4/5 at the other end.

 Good luck

 Tim Robinson
 Basingstoke, UK



 Gordon Henderson wrote:

 I have a client intersted in a system, but they have an ISDN30 line -
 the down side is that I've not done any before...

 Now, I've no reason to think it won't work, but as going on-site with
 a new card and no first-hand knowledge isn't particularly wise, I'm
 wondering about the best way to do some testing...

 So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30
 willing to let me play? ;-)

 Failing that, what test gear exists to pretend to be an exchange line?
 (Although I'm suspecting it's going to be outside my budget )-:

 But finally, can you run a TE110P card in master mode? ie. can I get
 2 of these, and put one in a separate box and use it to pretend to be
 the (BT) exchange, with the other box doing what it's supposed to do?

 If it is possible, what do I need in the way of cross-over cables, etc.?

 Thanks,

 Gordon
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Re: [asterisk-users] No sound with Playback() or Background()

2007-02-28 Thread Joanna Liza Mariazeta


Hi Jake,

Perhaps you can add

NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
FAILED or SUCCESS in the CLI

Hope that helps.

Best Regards,
Joanna

On 2/28/07, Kuba [EMAIL PROTECTED] wrote:


After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very
strange problem. There is no sound with Playback() or Background()
commands.

Even though, Asterisk console shows the file is being played when I call
the extension ( i.e. echo test), I can't hear anything.


My echo test extension looks like this:

exten = 600,1,Answer

exten = 600,2,Playback(demo-echotest)

exten = 600,3,Echo

exten = 600,4,Playback(demo-echodone)


exten = 600,5,Hangup

Console shows something like that when I call:

-- Executing Answer(SIP/206-081a7160, ) in new stack
-- Executing Playback(SIP/206-081a7160, demo-echotest) in new stack
-- Playing 'demo-echotest' (language 'en')

So it looks like Asterisk is playing the file, but I can't hear anything.

The files demo-echotest.gsm and demo-echodone.gsm are present in
/var/lib/asterisk/sounds, so this is not the matter of missing files.

The same problem occurs with every file I try to play with Playback() or
Background() commands.


Any ideas ?

Thanks
Jake





--
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  http://www.domeny.alpha.pl
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Re: [asterisk-users] ISDN30 testing questions ...

This guy could save his brain cells by just getting his shining good 'ol voip 
pabx box interface
directly with the existing pabx of his client.

I just wonder what is the brand name of that existing pabx?

 Gordon Henderson wrote:
 On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:

 Hi,

 What is the main purpose of this setup by the way?

 For me? To provde a client with a VoIP capable PBX in their office to
 replace their current steam driven PBX...

 (And hopefully to earn a few Â£Â£in the process!)

 I have a lot of installations now with pure analogue lines, now want
 to do one with a digital line, but I want to test it out before going
 on-site, so I don't look like an idiot when it doesn't work...

 Gordon
 You should be in for a treat then.  Digital is much easier IMO than
 analog setups.

 I have not touched an analog setup recently (about a year) but unless
 things have changed alot in echo cancellation and analog, that is what I
 spent most of my time doing, chasing intermittent echo.  I have yet to
 experience this with digital.  Not to say I have never encountered echo,
 I have, but after adjusted the echo is gone.

 Also, there is a certain thrill to watch 30 channels come up one after
 another in rapid succession.

 Thanks,
 Steve Totaro
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Re: [asterisk-users] Playback uses channel's language, background doesn't

2007-02-28 Thread Joanna Liza Mariazeta


Hi Cameron,

Why not automatically set the language that should be use at the beginning.
Set(LANGUAGE()=nz)

Hope that helps.

Best Regards,
Joanna

On 2/28/07, Moises Silva [EMAIL PROTECTED] wrote:


http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage

There you can found how you can get the current language ( the same
used by playback ), so you can set a local variable to the current
language and use it instead of the blank value

Regards

On 2/26/07, kjcsb [EMAIL PROTECTED] wrote:




 it may be a bug, try creating a simple test script with only 2
 extensions, one with playback the other one with background and see
 how it works, also post here the asterisk version you are using.
 Asterisk 1.2.13

 exten = 98765,1,Playback(to-listen-to-it)
 exten =
 98764,1,Background(to-listen-to-it|m||macro-systemrecording)
 exten = 98763,1,Background(to-listen-to-it)

 -- Executing Playback(SIP/112233-09289b40, to-listen-to-it) in
new
 stack
 -- Playing 'to-listen-to-it' (language 'nz')
 -- Executing Hangup(SIP/112233-09289b40, ) in new stack
   == Spawn extension (116-2000, h, 1) exited non-zero on
 'SIP/112233-09289b40'
 -- Executing BackGround(SIP/112233-09289b40,
 to-listen-to-it|m||macro-systemrecording) in new stack
 -- Playing 'to-listen-to-it' (language '')
 -- Executing Hangup(SIP/112233-09289b40, ) in new stack
   == Spawn extension (116-2000, h, 1) exited non-zero on
 'SIP/112233-09289b40'
 -- Executing BackGround(SIP/112233-09289b40,
 to-listen-to-it) in new stack
 -- Playing 'to-listen-to-it' (language 'nz')
 -- Executing Hangup(SIP/112233-09289b40, ) in new stack
   == Spawn extension (116-2000, h, 1) exited non-zero on
 'SIP/112233-09289b40'

 So it seems assume that since I passed a blank language override to the
 Background application, that I want a blank language. Any ideas on how
to
 get background to use the default language?

 Regards

 Cameron
  
  Inbox full of unwanted email? Get leading protection and 1GB storage
with
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RE: [asterisk-users] ISDN30 testing questions ...

2007-02-28 Thread asterisk

What is the make of the existing pabx? Be aware that if it is an older pabx
the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather than true isdn30e.

Neil

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: 28 February 2007 12:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ISDN30 testing questions ...

On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:

 Hi,

 What is the main purpose of this setup by the way?

For me? To provde a client with a VoIP capable PBX in their office to 
replace their current steam driven PBX...

(And hopefully to earn a few Â£Â£in the process!)

I have a lot of installations now with pure analogue lines, now want to do 
one with a digital line, but I want to test it out before going on-site, 
so I don't look like an idiot when it doesn't work...

Gordon


 Hi Gordon

 'Fraid I don't have a line you can 'play with' so to speak! However,
 firstly, I have installed several E1 based Asterisk systems both in UK
 and elsewhere, and apart from a few telco issues in a Latin American
 country, it just works.

 You can do as you suggest, i.e. have two Asterisk boxes back to back,
 and we have done this on many occasions for load-testing our systems. To
 do so you will need to make an E1 crossover with pins 1/2 going to pins
 4/5 at the other end.

 Good luck

 Tim Robinson
 Basingstoke, UK



 Gordon Henderson wrote:

 I have a client intersted in a system, but they have an ISDN30 line -
 the down side is that I've not done any before...

 Now, I've no reason to think it won't work, but as going on-site with
 a new card and no first-hand knowledge isn't particularly wise, I'm
 wondering about the best way to do some testing...

 So is there anyone in Devon/Cornwall (England!!!) with an ISDN 30
 willing to let me play? ;-)

 Failing that, what test gear exists to pretend to be an exchange line?
 (Although I'm suspecting it's going to be outside my budget )-:

 But finally, can you run a TE110P card in master mode? ie. can I get
 2 of these, and put one in a separate box and use it to pretend to be
 the (BT) exchange, with the other box doing what it's supposed to do?

 If it is possible, what do I need in the way of cross-over cables, etc.?

 Thanks,

 Gordon
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Re: [asterisk-users] No sound with Playback() or Background()

2007-02-28 Thread john beaman

Jake,
  Check to make sure you have the sound files for whatever audio format 
(gsm.wav, etc) that you are using.  I don't remember the details, but Asterisk 
quit including the sound files in the base distribution to minimize the size of 
the download.  Then, in a later version, they have a script that will prompt 
you for some info, then will download and install the sound files that you want 
to use.



John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331

 [EMAIL PROTECTED] 2/28/2007 7:30:03 AM 
Hi Jake,

Perhaps you can add

NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
FAILED or SUCCESS in the CLI

Hope that helps.

Best Regards,
Joanna

On 2/28/07, Kuba [EMAIL PROTECTED] wrote:

 After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very
 strange problem. There is no sound with Playback() or Background()
 commands.

 Even though, Asterisk console shows the file is being played when I call
 the extension ( i.e. echo test), I can't hear anything.


 My echo test extension looks like this:

 exten = 600,1,Answer

 exten = 600,2,Playback(demo-echotest)

 exten = 600,3,Echo

 exten = 600,4,Playback(demo-echodone)


 exten = 600,5,Hangup

 Console shows something like that when I call:

 -- Executing Answer(SIP/206-081a7160, ) in new stack
 -- Executing Playback(SIP/206-081a7160, demo-echotest) in new stack
 -- Playing 'demo-echotest' (language 'en')

 So it looks like Asterisk is playing the file, but I can't hear anything.

 The files demo-echotest.gsm and demo-echodone.gsm are present in
 /var/lib/asterisk/sounds, so this is not the matter of missing files.

 The same problem occurs with every file I try to play with Playback() or
 Background() commands.


 Any ideas ?

 Thanks
 Jake





 --
 --- Domeny w ULTRA NISKICH cenach: ---
 .pl - 29 zl, .com.pl - 22,50 zl, reg - 7,50 zl
   http://www.domeny.alpha.pl 
 --


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printing, distributing or use of this transmission is strictly

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Re: [asterisk-users] Voice mail is not giving unavailable or busy prompts

2007-02-28 Thread C F


If the temp message exists then that will play. The user has to log
into the mailbox (app_voicemailmain) and select 0 for mailbox options,
and delete the temp message. Or you could do it using the shell.


On 2/27/07, Stephen Bosch [EMAIL PROTECTED] wrote:

Hi:

This should be easy. I'm running 1.2.15.

When a caller calls someone's voice mail, it goes straight to a beep,
even though there is an unavail.wav file in that user's voice mail
directory.

Here is the relevant part of extensions.conf:

[internal]
exten = 2211,1,Dial(SIP/211,10)
exten = 2211,2,VoiceMail([EMAIL PROTECTED])
exten = 2211,3,Hangup

Here is the relevant part of voicemail.conf:

[default]
211 = ,Mr Test,[EMAIL PROTECTED]

Here's what I see in the console:

-- Executing Dial(SIP/210-081990b0, SIP/211|10) in new stack
 -- Called 211
 -- SIP/211-0819e5f0 is ringing
 -- Nobody picked up in 1 ms
 -- Executing VoiceMail(SIP/210-081990b0, [EMAIL PROTECTED]) in new 
stack
 -- Playing '/var/spool/asterisk/voicemail/default/211/unavail' (language 
'en')
 -- Playing 'vm-intro' (language 'en')
 -- Playing 'beep' (language 'en')
 -- Recording the message
 -- x=0, open writing:  
/var/spool/asterisk/voicemail/default/211/tmp/EWtUPC format: wav, 0x81a3c98
 -- User ended message by pressing #
 -- Playing 'auth-thankyou' (language 'en')
 -- Executing Hangup(SIP/210-081990b0, ) in new stack
   == Spawn extension (internal, 2211, 3) exited non-zero on 'SIP/210-081990b0'

This is what is actually in /var/spool/asterisk/voicemail/default/211:

 asterisk1 211 # ls -liah
 total 108K
 4918844 drwx-- 7 root root 4.0K Feb 27 17:59 .
 4898961 drwxr-xr-x 5 root root 4.0K Feb 27 17:05 ..
 4918846 drwx-- 2 root root 4.0K Feb 27 18:32 INBOX
 4918850 drwx-- 2 root root 4.0K Feb 27 17:12 Old
 4918849 -rwx-- 1 root root  56K Feb 27 17:10 busy.wav
 4918845 drwx-- 2 root root 4.0K Feb 27 17:05 temp
 4918847 drwx-- 2 root root 4.0K Feb 27 18:32 tmp
 4931585 drwxr-xr-x 2 root root 4.0K Feb 27 17:59 unavail
 4918848 -rwx-- 1 root root  20K Feb 27 17:13 unavail.wav

Asterisk creates that unavail directory after the first time someone
tries to call in.

Ideas?

-Stephen-
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Re: [asterisk-users] ISDN30 testing questions ...


[EMAIL PROTECTED] wrote:

And how would you be able to make a test telephone call with ISDN30 when you 
don't have an E1
link? You gonna have two asterisk box connected peer to peer and have the other 
as the master and
the other as a slave. Or the other way of saying that your other asterisk box 
generates the
signalling/framing while the other is the recipient? How would you be able to 
know for sure if
this works?

  
The bottom line is that it will work if you get it configured properly.  
Asterisk can be configured in many different E1 configurations.  Having 
a tech at the telco on the phone is helpful if thy can answer your 
questions such as Do you use CRC4 error correction?


Going through the motions of setting it up in a loopback test 
environment is a very good idea so you will at least have some 
experience and confidence.  You will certainly learn things that you 
didn't know, and will definitely be better off for it. 

The only way to know for sure it works is to make it work, and it will.  
That should be your attitude.


Thanks,
Steve Totaro
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RE: [asterisk-users] ISDN30 testing questions ...


On Wed, 28 Feb 2007, asterisk wrote:


What is the make of the existing pabx? Be aware that if it is an older pabx
the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather than true isdn30e.


This is another issue I didn't mention (saw no need!) Their ancient PBX 
does indeed use DASS2, but BT are killing this off at the end of July, and 
will be replacing it with proper ISDN2e, hence another need to get a new 
PBX in when their existing PBX is out of service and can't be upgraded...


Gordon
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Re: [asterisk-users] ISDN30 testing questions ...

2007-02-28 Thread Julian Lyndon-Smith


Gordon Henderson wrote:

On Wed, 28 Feb 2007, asterisk wrote:

What is the make of the existing pabx? Be aware that if it is an older 
pabx

the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather than true isdn30e.


This is another issue I didn't mention (saw no need!) Their ancient PBX 
does indeed use DASS2, but BT are killing this off at the end of July, 
and will be replacing it with proper ISDN2e, hence another need to get 


do you mean ISDN2e or ISDN30e ?

a new PBX in when their existing PBX is out of service and can't be 
upgraded...


Gordon
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Re: [asterisk-users] ISDN30 testing questions ...

The nature of the existing PABX to be interface is of prime importance. Before 
you say it will
works, please check the latest postings.

 [EMAIL PROTECTED] wrote:
 And how would you be able to make a test telephone call with ISDN30 when you 
 don't have an E1
 link? You gonna have two asterisk box connected peer to peer and have the 
 other as the master
 and
 the other as a slave. Or the other way of saying that your other asterisk 
 box generates the
 signalling/framing while the other is the recipient? How would you be able 
 to know for sure if
 this works?


 The bottom line is that it will work if you get it configured properly.
 Asterisk can be configured in many different E1 configurations.  Having
 a tech at the telco on the phone is helpful if thy can answer your
 questions such as Do you use CRC4 error correction?

 Going through the motions of setting it up in a loopback test
 environment is a very good idea so you will at least have some
 experience and confidence.  You will certainly learn things that you
 didn't know, and will definitely be better off for it.

 The only way to know for sure it works is to make it work, and it will.
 That should be your attitude.

 Thanks,
 Steve Totaro
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Re: [asterisk-users] ISDN30 testing questions ...


On Wed, 28 Feb 2007, Julian Lyndon-Smith wrote:


Gordon Henderson wrote:

On Wed, 28 Feb 2007, asterisk wrote:

What is the make of the existing pabx? Be aware that if it is an older 
pabx

the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather than true isdn30e.


This is another issue I didn't mention (saw no need!) Their ancient PBX 
does indeed use DASS2, but BT are killing this off at the end of July, and 
will be replacing it with proper ISDN2e, hence another need to get 


do you mean ISDN2e or ISDN30e ?


Yes, sorry. ISDN30e...

Thanks,

Gordon
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Re: [asterisk-users] ISDN30 testing questions ...


[EMAIL PROTECTED] wrote:

The nature of the existing PABX to be interface is of prime importance. Before 
you say it will
works, please check the latest postings.

  
But it isn't because they are changing protocols/signaling to something 
that is Asterisk compatible.  Please check the latest postings.

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Re: [asterisk-users] Saving Dialplan in CLI

2007-02-28 Thread John C. Wolosuk Jr.

a disable mode for reading general stuff and an enable mode for 
configuration related tasks I think would be a very nice feature fro 
asterisk to have. especially in this situation, some type of copy 
running-config startup-config would have proven useful. lucky for me my 
screw up wasn't on a production machine...


---
John C. Wolosuk Jr.
Unix/Linux Systems Administrator
Academic Computing  Communications Center
University of Illinois @ Chicago

E-Mail: jwolosuk at uic dot edu
---


Steve Totaro wrote:

Philipp Kempgen wrote:

John C. Wolosuk Jr. wrote:

 
Is there anyway to unset the extensions.conf definition of 
writeprotect=yes while in the CLI interface (or by other mechanism) 
to enable the dialplan save command? I accidentally overwrote my 
extensions.conf but still have a running copy of asterisk with the 
old dial plan running in memory.



show dialplan
might be your friend but the output is not an executable dialplan.

Regards,
  Philipp

  
A Ciscoesque show command, show running-configuration would be pretty 
cool. ___

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Re: [asterisk-users] ISDN30 testing questions ...

2007-02-28 Thread Tim Panton



On 28 Feb 2007, at 06:53, Gordon Henderson wrote:


On Wed, 28 Feb 2007, Julian Lyndon-Smith wrote:


Gordon Henderson wrote:

On Wed, 28 Feb 2007, asterisk wrote:
What is the make of the existing pabx? Be aware that if it is an  
older pabx

the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather than true isdn30e.
This is another issue I didn't mention (saw no need!) Their  
ancient PBX does indeed use DASS2, but BT are killing this off at  
the end of July, and will be replacing it with proper ISDN2e,  
hence another need to get


do you mean ISDN2e or ISDN30e ?


Yes, sorry. ISDN30e...


If you don't have a spare ISDN config to test on, I advise the  
following:
	1) design and test your overall dialplan/phones/routing on a test  
box with IAX (or SIP I suppose)
		connectivity to an ITSP - that way you can test DIDs , internal  
transfers, suitability of
		handsets etc. (You will find that the behavior of incoming calls on  
ISDN is more like that of
		IAX than it is of analog PSTN - all calls come in to an extension,  
hangup works properly, you get

callerID immediately, DTMF works etc)
	2) build you real server with the ISDN card in it and try and find  
somewhere to test it for an

hour just to check you have the zaptel and zapata config files sorted.
	3) when you do connect to the BT line, if it doesn't work call BT  
and ask them what they see.
quite often the problem is that they have marked the line as out-of- 
service and are waiting

for you to tell them to re-enable it :-)

Tim.


Thanks,

Gordon
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Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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RE: [asterisk-users] TE110P: Error == Asterisk died with code 1.

2007-02-28 Thread Jeronimo Romero

Thank you all. Was a signaling issue. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Wednesday, February 28, 2007 12:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] TE110P: Error == Asterisk died with code
1.

On Tue, Feb 27, 2007 at 08:01:41PM -0500, Jeronimo Romero wrote:
 Running Asterisk 1.2.9. I just installed a TE110P card and configured
 zaptel.conf  zapata.conf. The config files look right to me but I'm
 getting the following error when trying to start asterisk:

 Asterisk died with code 1.
 Automatically restarting Asterisk.

 Does anyone have any idea what is wrong with this configuration??
 Thanks in advance!!!

What is the output of:

  cat /proc/zaptel/*

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] No sound with Playback() or Background()

2007-02-28 Thread Kuba


Joanna Liza Mariazeta wrote:

Perhaps you can add

NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
FAILED or SUCCESS in the CLI


Hi Joanna,

I added that, but it looks like it does nothing :(. I don't see any 
status after Playback in the CLI.


All I get is:

-- Executing Answer(SIP/206-081af4c8, ) in new stack
-- Executing Playback(SIP/206-081af4c8, demo-echotest) in new stack
-- Playing 'demo-echotest' (language 'en')

Then, when I hang up

== Spawn extension (12ga, 600, 2) exited non-zero on 'SIP/206-081af4c8'


Regards
Jake



--
--- Domeny w ULTRA NISKICH cenach: ---
.pl - 29 zl, .com.pl - 22,50 zl, reg - 7,50 zl
 http://www.domeny.alpha.pl
--


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Re: [asterisk-users] No sound with Playback() or Background()

This can happen if you have a Digium card (maybe Sangoma too) in the 
system that is configured, but has no actual line plugged into it.  I 
don't know if this applies to analog, but I know it applies to T-1/PRI/E-1


Kuba wrote:

Joanna Liza Mariazeta wrote:

Perhaps you can add

NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
FAILED or SUCCESS in the CLI


Hi Joanna,

I added that, but it looks like it does nothing :(. I don't see any 
status after Playback in the CLI.


All I get is:

-- Executing Answer(SIP/206-081af4c8, ) in new stack
-- Executing Playback(SIP/206-081af4c8, demo-echotest) in new stack
-- Playing 'demo-echotest' (language 'en')

Then, when I hang up

== Spawn extension (12ga, 600, 2) exited non-zero on 'SIP/206-081af4c8'

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[asterisk-users] Timing, use analog card, ZT Dummy etc.

2007-02-28 Thread voiplist


Hello, we are setting up another system that will run either 1.2.4,
the latest version of 1.2 or 1.4. We have not yet decided on the
version.

Anyhow, this is a higher volume system (dual processor) which will
handle 30-50 simultaneous calls with 60 to 100 simultaneous channels
lit up. Most calls are g711 with very little g729 and a little gsm
mixed in.

We have a similar system doing exactly this, quite well.

With our existing system we have a single span Digium T1 card
installed, which we never ended up using. Nice it is in there though
because Asterisk uses it for timing.

The new system will be pure IP with no need for Analog or T1 circuits.

Questions are:

1- Can I really get away with using ZT Dummy on a high volume system
like this and put no card in?

2- If I can, should I even risk it or just put a card in?

3- I obviously don't want to put a $500 T1 card in but I do have a
Digium Analog card with 2 FXO modules. I also have some clone cards.
The question is, should I use the clone cards and will they work
reliably just for timing. OR should I use the Digium card?

4- If I use the Digium card in, do I need to also waste a module or
can I just put the bare card in with no modules since it's just for
timing?

Thanks for any help, I will be moving forward today more than likely
and thought I would get a little advice from the list first.
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[asterisk-users] h323 how to set it up?

2007-02-28 Thread Florea Igor

Hi all,
I have some questions about h323. Is it mandatory to install a oh323 or I can 
do h323 calls without patching or adding any new drivers ti asterisk?
I did compile the asterisk with channel driver chan_h323 but it still gives me 
this error:
[Feb 28 18:12:58] WARNING[1902]: app_dial.c:1081 dial_exec_full: Unable to 
create channel of type 'H323' (cause 66 - Channel not implemented)
what shoul I do to have it implemented?
Can somebody recommend some references on how to set up h323 ?
Thx,
Igor

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[asterisk-users] multiple phones registered for the same user

2007-02-28 Thread Ricardo Carvalho


Dear all,

I've noticed that when I have a phone registered in Asterisk, and then I 
register another phone with the same user, the sip show peers in the 
CLI shows that Asterisk replaced the IP of the first phone by the IP of 
the last one registered for that user. Consequently, if someone calls 
that user, only the last phone rings!!
How may I configure Asterisk to be able to fork all incoming calls to 
every phones registered for each user, so that every phone ring until 
someone answers the call in one of them?


Thanks in advance,
Ricardo.
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[asterisk-users] about bluetooth channel

2007-02-28 Thread Iban Lopetegi Zinkunegi


28th February

I am working with Asterisk 1.2.15. I have configured sip.conf for two soft 
phones (I am using Xlite).I have installed the Bluez stack and so far, i 
manage to make a phone call from a soft phone to a GSM network. However, i 
have an audio problem. The soft phone can be heart by the GSM costumer but 
the voice in  Xlite is not transmitted to the GSM. In asterisk all i got is 
the next lines:


-- Executing Dial(SIP/Jack-081e39b0, BLT/nokia/07863342772) in new stack
[AG]  nokia  ATD07863342772;
   -- Called nokia
[AG]  nokia  OK
[AG]  nokia  +CIEV: 3,2
[AG]  nokia  +CIEV: 4,2
[AG]  nokia  +CIEV: 3,3
   -- BLT/nokia is ringing
[AG]  nokia  +CIEV: 4,3
[AG]  nokia  +CIEV: 1,1
   -- BLT/nokia answered SIP/Jack-081e39b0
Feb 22 14:48:10 WARNING[5473]: /usr/src/bt/chan_bluetooth.c:622 sco_thread: 
SCO

thread started on fd 38, pid 5445
[AG]  nokia  +CIEV: 3,0
[AG]  nokia  +CIEV: 4,0
Feb 22 14:48:20 NOTICE[5470]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126 
rece   ived
Feb 22 14:48:31 NOTICE[5470]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126 
rece   ived
Feb 22 14:48:41 NOTICE[5470]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126 
rece   ived
Feb 22 14:48:52 NOTICE[5470]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126 
rece   ived

[AG]  nokia  ATH
[AG]  nokia  AT+CHUP
 == Spawn extension (internal, 007863342772, 1) exited non-zero on 
'SIP/Jack-08   
1e39b0'


Do you know where could be the problem?

_
Acepta el reto MSN Premium: Protección para tus hijos en internet. 
Descárgalo y pruébalo 2 meses gratis. 
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RE: [asterisk-users] multiple phones registered for the same user

2007-02-28 Thread Azfhasterisk

Create a different user for each phone and create a ring group with the
phones that you want to ring.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Wednesday, February 28, 2007 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] multiple phones registered for the same user

Dear all,

I've noticed that when I have a phone registered in Asterisk, and then I 
register another phone with the same user, the sip show peers in the 
CLI shows that Asterisk replaced the IP of the first phone by the IP of 
the last one registered for that user. Consequently, if someone calls 
that user, only the last phone rings!!
How may I configure Asterisk to be able to fork all incoming calls to 
every phones registered for each user, so that every phone ring until 
someone answers the call in one of them?

Thanks in advance,
Ricardo.
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Re: [asterisk-users] multiple phones registered for the same user


Ricardo Carvalho wrote:

Dear all,

I've noticed that when I have a phone registered in Asterisk, and then I 
register another phone with the same user, the sip show peers in the 
CLI shows that Asterisk replaced the IP of the first phone by the IP of 
the last one registered for that user. Consequently, if someone calls 
that user, only the last phone rings!!
How may I configure Asterisk to be able to fork all incoming calls to 
every phones registered for each user, so that every phone ring until 
someone answers the call in one of them?


You can't have more than 1 device registered to a SIP account in 
Asterisk.  Have the phones register as different SIP User IDs, then use 
something like Dial(SIP/user1SIP/user2SIP/user3) to ring all phones at 
once.  This is covered over and over in the mailing list archives.

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Re: [asterisk-users] h323 how to set it up?

2007-02-28 Thread Rodrigo Gonzalez


Florea Igor wrote:

Hi all,
I have some questions about h323. Is it mandatory to install a oh323 or I can 
do h323 calls without patching or adding any new drivers ti asterisk?
I did compile the asterisk with channel driver chan_h323 but it still gives me 
this error:
[Feb 28 18:12:58] WARNING[1902]: app_dial.c:1081 dial_exec_full: Unable to 
create channel of type 'H323' (cause 66 - Channel not implemented)

what shoul I do to have it implemented?
Can somebody recommend some references on how to set up h323 ?
Thx,
Igor

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 Read README file in channels/h323
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Re: [asterisk-users] multiple phones registered for the same user

2007-02-28 Thread Ricardo Carvalho

Can't I register multiple phones with the same user/password? That's 
what I pretend to do, not ring groups...


Thanks,
Ricardo.




Azfhasterisk wrote:

Create a different user for each phone and create a ring group with the
phones that you want to ring.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Wednesday, February 28, 2007 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] multiple phones registered for the same user

Dear all,

I've noticed that when I have a phone registered in Asterisk, and then I 
register another phone with the same user, the sip show peers in the 
CLI shows that Asterisk replaced the IP of the first phone by the IP of 
the last one registered for that user. Consequently, if someone calls 
that user, only the last phone rings!!
How may I configure Asterisk to be able to fork all incoming calls to 
every phones registered for each user, so that every phone ring until 
someone answers the call in one of them?


Thanks in advance,
Ricardo.
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[asterisk-users] Registrations, how many is too many?

2007-02-28 Thread voiplist


Anyone have any idea if there is some sort of limitation to the number
of SIP or IAX end points which can register to an Asterisk system
(2.8Ghz dual processor, 2GB ram) while also handling 30-50
simultaneous calls without getting into trouble?

Of course the 30-50 simultaneous calls end up being 60-100 channels of
mostly G711 VoIP.

We have seen issues where our Asterisk just gets all crazy and SIP
quits working all together, to the point sometimes where we can't even
fix it with a restart.

At one point we were using all realtime for IAX and SIP clients, then
we went to text files (more or less), still we are seeing this issue.

When this happens we can't even do simple things with SIP like sip
show peers etc. because Asterisk just says that the application
doesn't exist.

This has been a battle for a few months and we can't put our finger on
it. Can't seem to figure out when it's going to happen either which is
VERY tough on the nerves to say the least.

This happens during peak times but also in the middle of the night
when call volume is slow to non existent.

The only thing that's constant during both peak and non peak times is
the amount of registrations the system deals with.

We have approx 1500-1800 end points registering to this particular
system at any one time. This is a split between IAX and SIP not sure
what the percentage of each is at the moment.

It's been a long time since a problem has beat me/us and this one has
won so far.

Any help in getting my sanity back would be REALLY appreciated.
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[asterisk-users] seeing DTMF passed to Voicemail

2007-02-28 Thread cb

I'm having a strange issue. My voicemail is working fine, however,  
any time I try to access it via one of my analog phones that are  
connecting to Asterisk via a Mediatrix 1124... the voicemail system  
complains I've entered the wrong password.


There is about a 15 second pause between when I finish dialing in the  
password, and it complains it is wrong.


This ONLY happens with phones connected via the Mediatrix. My IP  
phones, and soft phones all work fine and have no problems accessing  
voicemail. And I know the VM accounts related to the Mediatrix based  
phones are ok, as I can access them via an IP phone dialing into  
general VM access (and then specifying the box and password from there).


I'm guessing that the Mediatrix is failing to send the DTMF tones  
correctly, or possibly send them at all. I have it set to use  
RFC-2833, same as my IP phones.


Is there somewhere or someway to see in Asterisk either via a debug  
command, or in some log somewhere, what the VoiceMail system thinks  
is being entered? Or, has anyone else run into something similar and  
knows why it keeps rejecting passwords sent via the Mediatrix?


-chris
www.mythtech.net


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Re: [asterisk-users] multiple phones registered for the same user


Ricardo Carvalho wrote:
Can't I register multiple phones with the same user/password? That's 
what I pretend to do, not ring groups...


No, you cannot register multiple phones with the same user/password.
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Re: [asterisk-users] TE110P: Error == Asterisk died with code 1.

2007-02-28 Thread Tzafrir Cohen

On Wed, Feb 28, 2007 at 10:47:48AM -0500, Jeronimo Romero wrote:
 Thank you all. Was a signaling issue. 

And for the benefit of those who will read the archive: how have you
debugged it? how have you resolved it?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Timing, use analog card, ZT Dummy etc.

2007-02-28 Thread Tzafrir Cohen

On Wed, Feb 28, 2007 at 10:00:10AM -0600, voiplist wrote:
 Hello, we are setting up another system that will run either 1.2.4,
 the latest version of 1.2 or 1.4. We have not yet decided on the
 version.
 
 Anyhow, this is a higher volume system (dual processor) which will
 handle 30-50 simultaneous calls with 60 to 100 simultaneous channels
 lit up. Most calls are g711 with very little g729 and a little gsm
 mixed in.
 
 We have a similar system doing exactly this, quite well.
 
 With our existing system we have a single span Digium T1 card
 installed, which we never ended up using. Nice it is in there though
 because Asterisk uses it for timing.
 
 The new system will be pure IP with no need for Analog or T1 circuits.
 
 Questions are:
 
 1- Can I really get away with using ZT Dummy on a high volume system
 like this and put no card in?

Currently A zaptel hardware source is more accurate than ztdummy, AFAIK.
But I don't have any good data on this.

Note that you don't have to have any channel defined in zapata.conf for
a card to be used as a zaptel timing source: just load the module and
probably a proper /etc/zaptel.conf that claims that this is a span that
does not use external timing.

 
 2- If I can, should I even risk it or just put a card in?
 
 3- I obviously don't want to put a $500 T1 card in but I do have a
 Digium Analog card with 2 FXO modules. I also have some clone cards.
 The question is, should I use the clone cards and will they work
 reliably just for timing. OR should I use the Digium card?

AFAIK, a 10$ X100P from eBay will be just as good for this purpose
(timing only!). Maybe also a HFC ISDN card using ZapBRI (again: you only
need the zaptel patch of bristuff, which is small and simple, not all of
the huge bristuffed asterisk / libpri) or mISDN and the external zaptel 
timing source patch.

 
 4- If I use the Digium card in, do I need to also waste a module or
 can I just put the bare card in with no modules since it's just for
 timing?

Haven't tested it.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] about bluetooth channel


Iban Lopetegi Zinkunegi wrote:

28th February

I am working with Asterisk 1.2.15. I have configured sip.conf for two 
soft phones (I am using Xlite).I have installed the Bluez stack and so 
far, i manage to make a phone call from a soft phone to a GSM network. 
However, i have an audio problem. The soft phone can be heart by the 
GSM costumer but the voice in  Xlite is not transmitted to the GSM. In 
asterisk all i got is the next lines:

I thought chan_bluetooth only worked with 1.4 head?

Thanks,
Steve
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[asterisk-users] Changing from email address for vociemail.conf

2007-02-28 Thread Dovid B

Hi List,
I put this in to my voicemail.conf as per the wikki and the users are still 
getting the emails from the root account. Any ideas on what it can be ?
I have: mailcmd=/usr/sbin/sendmail -v -t -f [EMAIL PROTECTED]
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Re: [asterisk-users] Registrations, how many is too many?


voiplist wrote:

Anyone have any idea if there is some sort of limitation to the number
of SIP or IAX end points which can register to an Asterisk system
(2.8Ghz dual processor, 2GB ram) while also handling 30-50
simultaneous calls without getting into trouble?

Of course the 30-50 simultaneous calls end up being 60-100 channels of
mostly G711 VoIP.

We have seen issues where our Asterisk just gets all crazy and SIP
quits working all together, to the point sometimes where we can't even
fix it with a restart.

At one point we were using all realtime for IAX and SIP clients, then
we went to text files (more or less), still we are seeing this issue.

When this happens we can't even do simple things with SIP like sip
show peers etc. because Asterisk just says that the application
doesn't exist.

This has been a battle for a few months and we can't put our finger on
it. Can't seem to figure out when it's going to happen either which is
VERY tough on the nerves to say the least.

This happens during peak times but also in the middle of the night
when call volume is slow to non existent.

The only thing that's constant during both peak and non peak times is
the amount of registrations the system deals with.

We have approx 1500-1800 end points registering to this particular
system at any one time. This is a split between IAX and SIP not sure
what the percentage of each is at the moment.

It's been a long time since a problem has beat me/us and this one has
won so far.

Any help in getting my sanity back would be REALLY appreciated.

Time for SER?

Thanks,
Steve
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Re: [asterisk-users] multiple phones registered for the same user

2007-02-28 Thread Ricardo Carvalho


Too bad... Thanks for all replays.

Regards,
Ricardo.





Eric ManxPower Wieling wrote:

Ricardo Carvalho wrote:
Can't I register multiple phones with the same user/password? That's 
what I pretend to do, not ring groups...


No, you cannot register multiple phones with the same user/password.



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Re: [asterisk-users] Changing from email address for vociemail.conf

2007-02-28 Thread Jacob Helwig

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

; Who the e-mail notification should appear to come from
;[EMAIL PROTECTED]


Dovid B wrote:
 Hi List,
 I put this in to my voicemail.conf as per the wikki and the users are
 still getting the emails from the root account. Any ideas on what it can
 be ?
 I have: mailcmd=/usr/sbin/sendmail -v -t -f [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
  
 
 
 
 
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-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFF5b3ARhLSniguQyERAtrfAJ9XTsPRtgk/yxV/NivK36YgHvu7mQCdGMuU
awieqQ/FhGDyBZ4aAKjKioc=
=IIii
-END PGP SIGNATURE-
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Re: [asterisk-users] No sound with Playback() or Background()

2007-02-28 Thread Doug Garstang

I goofed that up on my dCAP exam. Spent 20 valuable minutes trying to 
fix it!


Eric ManxPower Wieling wrote:
This can happen if you have a Digium card (maybe Sangoma too) in the 
system that is configured, but has no actual line plugged into it.  I 
don't know if this applies to analog, but I know it applies to 
T-1/PRI/E-1


Kuba wrote:

Joanna Liza Mariazeta wrote:

Perhaps you can add

NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
FAILED or SUCCESS in the CLI


Hi Joanna,

I added that, but it looks like it does nothing :(. I don't see any 
status after Playback in the CLI.


All I get is:

-- Executing Answer(SIP/206-081af4c8, ) in new stack
-- Executing Playback(SIP/206-081af4c8, demo-echotest) in new stack
-- Playing 'demo-echotest' (language 'en')

Then, when I hang up

== Spawn extension (12ga, 600, 2) exited non-zero on 'SIP/206-081af4c8'

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RE: [asterisk-users] chan_sip.c:10173 handle_response: Dont knowhowtohandle a 202 Accepted respons

2007-02-28 Thread Bala Neelakantan

Yuan,

It looks like you are getting 202 for SIP Request method MESSAGE.  The 202
response is processed properly.  Need to see the message fully.  You can
capture sip debug if you don't have ethereal.  This will provide more detail
call flow.

.{N/MESSAGE sip:[EMAIL PROTECTED] SIP/2.0

Via: SIP/2.0/U
13:42:12.761685 IP 10.0.0.10.5060  10.0.0.201.5060: UDP, length: 468
E:[EMAIL PROTECTED]
..

...SIP/2.0 202 Accepted

Via: SIP/2.0/UDP 10.0.0.201:5060
13:42:12.793347 IP 10.0.0.10.5060  10.0.0.201.5060: UDP, length: 399
E;[EMAIL PROTECTED]
..


Thanks,
Neel


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Tuesday, February 27, 2007 4:23 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] chan_sip.c:10173 handle_response: Dont
knowhowtohandle a 202 Accepted respons

From: Bala Neelakantan [EMAIL PROTECTED]
Date: Tue, 27 Feb 2007 14:21:32 -0600

Looks like asterisk is receiving 202 while it is not expecting it.

/*! \brief Handle SIP response in dialogue */
/* XXX only called by handle_request */
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct
sip_request *req, int ignore, int seqno)

Can you provide ethereal capture when you see this log message?

Neel,

Thanks for the reply.  I don't have ethereal on the machine and not sure how

to capture - non-graphic terminal environment.  Below is output from 
tcpdump.  In this session, I see two 202 Accepted from 1.4.0, only one 
don't know notice.  Interestingly, identical tests between two 1.2.13 
Asterisk does not produce this.

I assume that this is nothing serious, because the session completes without

any problem, and the message is only a notice.  If anything, I'll simply 
revert to 1.2. (These are non-production.)

Yuan Liu

13:42:12.685850 IP 10.0.0.10.5060  10.0.0.201.5060: UDP, length: 749
E.. [EMAIL PROTECTED]
..

...INVITE sip:[EMAIL PROTECTED] SIP/2.0

Via: SIP/2.0/UDP 1
13:42:12.686783 IP 10.0.0.201.5060  10.0.0.10.5060: UDP, length: 430
E...D[EMAIL PROTECTED] +
...
..
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.0.0.10:5060;br
13:42:12.687705 IP 10.0.0.201.5060  10.0.0.10.5060: UDP, length: 710
E...D'[EMAIL PROTECTED]
...
..
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.0.0.10:5060;branch
13:42:12.688229 IP 10.0.0.10.5060  10.0.0.201.5060: UDP, length: 363
[EMAIL PROTECTED]
..

s.WACK sip:[EMAIL PROTECTED] SIP/2.0
V
ia: SIP/2.0/UDP 10.0
13:42:12.761105 IP 10.0.0.201.5060  10.0.0.10.5060: UDP, length: 371
E...D([EMAIL PROTECTED] d
...
..
.{N/MESSAGE sip:[EMAIL PROTECTED] SIP/2.0

Via: SIP/2.0/U
13:42:12.761685 IP 10.0.0.10.5060  10.0.0.201.5060: UDP, length: 468
E:[EMAIL PROTECTED]
..

...SIP/2.0 202 Accepted

Via: SIP/2.0/UDP 10.0.0.201:5060
13:42:12.793347 IP 10.0.0.10.5060  10.0.0.201.5060: UDP, length: 399
E;[EMAIL PROTECTED]
..

..{MESSAGE sip:[EMAIL PROTECTED] SIP/2.0
V
ia: SIP/2.0/UDP
13:42:12.793863 IP 10.0.0.201.5060  10.0.0.10.5060: UDP, length: 448
E...D)[EMAIL PROTECTED] .
...
..
...oSIP/2.0 202 Accepted

Via: SIP/2.0/UDP 10.0.0.10:5060;
13:42:12.796133 IP 10.0.0.201.5060  10.0.0.10.5060: UDP, length: 332
[EMAIL PROTECTED] .
...
..
.Tw.BYE sip:[EMAIL PROTECTED] SIP/2.0

Via: SIP/2.0/UDP 1
13:42:12.796777 IP 10.0.0.10.5060  10.0.0.201.5060: UDP, length: 463
E[EMAIL PROTECTED]
..

...SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.0.0.201:5060;branc


Thanks,
Neel

-Original Message-

What does this mean?  Asterisk 1.2.13 talking to 1.4.0. (response from
1.4.0.)

Yuan Liu


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Re: [asterisk-users] Registrations, how many is too many?

2007-02-28 Thread voiplist


That only helps on the SIP side.. :(

Although it would help some..

Before we go making changes, we are really just trying to determine
what the cause is.



On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote:

voiplist wrote:
 Anyone have any idea if there is some sort of limitation to the number
 of SIP or IAX end points which can register to an Asterisk system
 (2.8Ghz dual processor, 2GB ram) while also handling 30-50
 simultaneous calls without getting into trouble?

 Of course the 30-50 simultaneous calls end up being 60-100 channels of
 mostly G711 VoIP.

 We have seen issues where our Asterisk just gets all crazy and SIP
 quits working all together, to the point sometimes where we can't even
 fix it with a restart.

 At one point we were using all realtime for IAX and SIP clients, then
 we went to text files (more or less), still we are seeing this issue.

 When this happens we can't even do simple things with SIP like sip
 show peers etc. because Asterisk just says that the application
 doesn't exist.

 This has been a battle for a few months and we can't put our finger on
 it. Can't seem to figure out when it's going to happen either which is
 VERY tough on the nerves to say the least.

 This happens during peak times but also in the middle of the night
 when call volume is slow to non existent.

 The only thing that's constant during both peak and non peak times is
 the amount of registrations the system deals with.

 We have approx 1500-1800 end points registering to this particular
 system at any one time. This is a split between IAX and SIP not sure
 what the percentage of each is at the moment.

 It's been a long time since a problem has beat me/us and this one has
 won so far.

 Any help in getting my sanity back would be REALLY appreciated.
Time for SER?

Thanks,
Steve
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RE: [asterisk-users] Registrations, how many is too many?

2007-02-28 Thread Yuan LIU

From: voiplist [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 10:54:30 -0600

Anyone have any idea if there is some sort of limitation to the number
of SIP or IAX end points which can register to an Asterisk system
(2.8Ghz dual processor, 2GB ram) while also handling 30-50
simultaneous calls without getting into trouble?

Of course the 30-50 simultaneous calls end up being 60-100 channels of
mostly G711 VoIP.

We have seen issues where our Asterisk just gets all crazy and SIP
quits working all together, to the point sometimes where we can't even
fix it with a restart.

At one point we were using all realtime for IAX and SIP clients, then
we went to text files (more or less), still we are seeing this issue.

When this happens we can't even do simple things with SIP like sip
show peers etc. because Asterisk just says that the application
doesn't exist.

This has been a battle for a few months and we can't put our finger on
it. Can't seem to figure out when it's going to happen either which is
VERY tough on the nerves to say the least.

This happens during peak times but also in the middle of the night
when call volume is slow to non existent.

The only thing that's constant during both peak and non peak times is
the amount of registrations the system deals with.

What kind of system stats do you see at different periods of time? (e.g., 
load average, network I/O, disk I/O, # of processes.)  When Asterisk says 
the application doesn't exist, is there any system errors at the time?  
What does top say?

Such info can give other people a better idea in order to do hair cut by 
E-mail.  When archived, they also help other people with similar problems.

Yuan Liu

We have approx 1500-1800 end points registering to this particular
system at any one time. This is a split between IAX and SIP not sure
what the percentage of each is at the moment.

It's been a long time since a problem has beat me/us and this one has
won so far.

Any help in getting my sanity back would be REALLY appreciated.

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[asterisk-users] Run-away Asterisk

2007-02-28 Thread Yuan LIU

After testing some AGI's, I noticed several extra Asterisk processes.  They 
are not zombies, but can't be killed by safe_asterisk.  Nor will they die 
when CLI issues stop now.  Then I read that each AGI spawns a separate 
Asterisk process.  But all my AGI calls have apparently completed 
successfully.  So there should be no reason for them to hang there.


Several questions:

1) Under what conditions will an AGI hang a process? (My test scripts are 
pretty simple, almost directly derived from agi-test.agi.)


2) How to detect run-away processes under 2.4 kernels?  In this kernel, each 
thread clusters process space and it's very difficult to distinguish them 
without killing the main process.


3) Any practical way to detect them from inside Asterisk - e.g., do some 
check after each AGI call?  All my AGISTATUS reports success.  I could use 
System() but isn't that cumbersome?


Thank you.

Yuan Liu


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Re: [asterisk-users] multiple phones registered for the same user

Yuan LIU wrote:

From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 10:57:43 -0600

Ricardo Carvalho wrote:
Can't I register multiple phones with the same user/password? That's 
what I pretend to do, not ring groups...

Ricardo,

Any particular reason for not using ring groups?

No, you cannot register multiple phones with the same user/password.

Just curious: can I register multiple phones with one user name but 
different passwords?

no.
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Re: [asterisk-users] Registrations, how many is too many?

2007-02-28 Thread C F


you using dynamic dns?

On 2/28/07, voiplist [EMAIL PROTECTED] wrote:

That only helps on the SIP side.. :(

Although it would help some..

Before we go making changes, we are really just trying to determine
what the cause is.



On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
 voiplist wrote:
  Anyone have any idea if there is some sort of limitation to the number
  of SIP or IAX end points which can register to an Asterisk system
  (2.8Ghz dual processor, 2GB ram) while also handling 30-50
  simultaneous calls without getting into trouble?
 
  Of course the 30-50 simultaneous calls end up being 60-100 channels of
  mostly G711 VoIP.
 
  We have seen issues where our Asterisk just gets all crazy and SIP
  quits working all together, to the point sometimes where we can't even
  fix it with a restart.
 
  At one point we were using all realtime for IAX and SIP clients, then
  we went to text files (more or less), still we are seeing this issue.
 
  When this happens we can't even do simple things with SIP like sip
  show peers etc. because Asterisk just says that the application
  doesn't exist.
 
  This has been a battle for a few months and we can't put our finger on
  it. Can't seem to figure out when it's going to happen either which is
  VERY tough on the nerves to say the least.
 
  This happens during peak times but also in the middle of the night
  when call volume is slow to non existent.
 
  The only thing that's constant during both peak and non peak times is
  the amount of registrations the system deals with.
 
  We have approx 1500-1800 end points registering to this particular
  system at any one time. This is a split between IAX and SIP not sure
  what the percentage of each is at the moment.
 
  It's been a long time since a problem has beat me/us and this one has
  won so far.
 
  Any help in getting my sanity back would be REALLY appreciated.
 Time for SER?

 Thanks,
 Steve
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Re: [asterisk-users] Registrations, how many is too many?

2007-02-28 Thread C F


any dns in the sip channel could do this not only dynamic

On 2/28/07, voiplist [EMAIL PROTECTED] wrote:

That only helps on the SIP side.. :(

Although it would help some..

Before we go making changes, we are really just trying to determine
what the cause is.



On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
 voiplist wrote:
  Anyone have any idea if there is some sort of limitation to the number
  of SIP or IAX end points which can register to an Asterisk system
  (2.8Ghz dual processor, 2GB ram) while also handling 30-50
  simultaneous calls without getting into trouble?
 
  Of course the 30-50 simultaneous calls end up being 60-100 channels of
  mostly G711 VoIP.
 
  We have seen issues where our Asterisk just gets all crazy and SIP
  quits working all together, to the point sometimes where we can't even
  fix it with a restart.
 
  At one point we were using all realtime for IAX and SIP clients, then
  we went to text files (more or less), still we are seeing this issue.
 
  When this happens we can't even do simple things with SIP like sip
  show peers etc. because Asterisk just says that the application
  doesn't exist.
 
  This has been a battle for a few months and we can't put our finger on
  it. Can't seem to figure out when it's going to happen either which is
  VERY tough on the nerves to say the least.
 
  This happens during peak times but also in the middle of the night
  when call volume is slow to non existent.
 
  The only thing that's constant during both peak and non peak times is
  the amount of registrations the system deals with.
 
  We have approx 1500-1800 end points registering to this particular
  system at any one time. This is a split between IAX and SIP not sure
  what the percentage of each is at the moment.
 
  It's been a long time since a problem has beat me/us and this one has
  won so far.
 
  Any help in getting my sanity back would be REALLY appreciated.
 Time for SER?

 Thanks,
 Steve
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Re: [asterisk-users] Long call setup times on SIP to zaptel calls

When we do SIP - SIP with asterisk 1.2, we do NOT experience this. 
Polycom 501s, the instant you hit Send on the phone or the digit map 
times out, the target phone rings AND you hear ringback.  it's instant, 
so I would guess this would be configuration on your end. back to the 
digit map timeout maybe?


 Watching the CLI it does look like it takes a long time for the
 channel to pick up an dial.
I'm a little confused about your 'pick up and dial' phrase because, 
well, we're talking about SIP here and not Zap :)


David Thomas wrote:

On 2/15/07, Jordan Novak [EMAIL PROTECTED] wrote:


I have had a lot of complaints about the time it takes to setup a call. I
have timed it and it is almost five seconds before it even starts 
ringing.
The SIP device sends the request almost instantly but the channel is 
taking
a long time to pickup and dial. It wouldn't be so bad but they hear 
nothing.
I would like to provide ringback before the zaptel actually starts 
ringing

the channel. Has anybody done this, it seems like it would be a zaptel
option.

Jordan Novak


I'm not sure if it's related, but we are doing only SIP to SIP calling
with Asterisk 1.4 and experience the same thing. The signaling shows
up instantly, but it takes 5-7 seconds before ringback is heard.
Watching the CLI it does look like it takes a long time for the
channel to pick up an dial.

regards,
David
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Re: [asterisk-users] Long call setup times on SIP to zaptel calls




Yuan LIU wrote:
Doesn't seem to happen in TDM400P and X100P cards, though.  Could it be 
some feature configured in your particular card?

Notice just after my name:
P.S. Polycom Soundpoint 501, *TDM* w/ 4xFXO, Asterisk 1.2.13

As El ManxPower mentioned, have you tried using ZapBarge to detect this? 
  That's the only way I could tell it was happening.


Moj




Yuan Liu

Doesn't matter how many numbers I want to send out the ZAP channel, 
this always seems to happen.


Asterisk 1.2.x is affected for sure.  I haven't tested 1.4 yet.  But 
if we could get this figured out, that would shave two seconds off MY 
nearly-five-second setup time.


Mojo





Jordan Novak wrote:
I have had a lot of complaints about the time it takes to setup a 
call. I have timed it and it is almost five seconds before it even 
starts ringing. The SIP device sends the request almost instantly but 
the channel is taking a long time to pickup and dial. It wouldn't be 
so bad but they hear nothing. I would like to provide ringback before 
the zaptel actually starts ringing the channel. Has anybody done 
this, it seems like it would be a zaptel option.

 Jordan Novak



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Re: [asterisk-users] Transfer Caller ID

As far as I can tell, the only way to do this using Polycom soundpoint 
phones and NOT asterisk's built-in blindxfer function, is to hit their 
Transfer button first, and then the Blind softkey that appears on the 
screen.  Then continue as normal; dial the number and hit Send I 
believe.  If you can get your operator to hit Transfer and then Blind 
automagically, this will work.  But if s/he wants to consult with the 
internal employee first, she would have to:


1   place caller on hold (phone button)
2   dial internal employee, talk, hangup
3   unhold caller
4   blind xfer caller to internal employee

Moj


Rob Schall wrote:

I probably have a screened transfer setup. Is that just a setting
somewhere I can easily change? I'm trying to avoid making users press
extra keys, like #1 or anything like that.

Rob




Jerry Jones wrote:

Not sure about others, but on Polycoms a blind transfer sends original
callerid, screened sends operators callerid


On Feb 19, 2007, at 8:55 AM, Rob Schall wrote:


I'm sure this was asked before, but I can't seem to make this work...

If a customer dials one of our DIDs, and the operator transfers that
call to another employee, the Caller ID doesn't seem to do what I would
expect it to. I would expect it to show the original caller's ID.

Example:
John calls in from the outside using (213-555-1234) and he calls into
the asterisk system (actually the operator).
The operator (a real person) answers the call and presses transfer on
her polycom 501 phone. I see an incoming call From: Operator. After I
pick up her call, she presses transfer one final time to complete the
transfer. However, now that the call has been completed, it still shows
From: Operator. I need it to show From: 213-555-1234.

I tried setting the o setting in Dial, but that didn't seem to fix
anything.

Rob

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Re: [asterisk-users] Limit on SIP phones on one server

But just to handle 10 simultaneous calls, you probably don't even need 1 
GHz!


Matt Richards wrote:

I don't see any reason why a single server wont handle 700 phones as
long as its powerful enough.
I would think that anything over 1GHz should be fine maybe less :)

Matty.

Jerry Geis wrote:

I have an application where I might need 700 SIP phones (wireless)
connected to one asterisk server. Will it do this?

The situation:

Only a small number (less than 10) will actually be talking at one time.

I presume asterisk can handle 700 SIP definitions correct?

Do I need to recompile anything to handle that many phones?

Thanks,

Jerry



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Re: [asterisk-users] Best FXO Gateway

2007-02-28 Thread kjcsb

Linksys SPA400 is a 4 port FXO gateway.

Cameron





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Re: [asterisk-users] multiple phones registered for the same user

2007-02-28 Thread Bayrouni

Eric ManxPower Wieling a écrit :
 Yuan LIU wrote:
 From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
 Date: Wed, 28 Feb 2007 10:57:43 -0600

 Ricardo Carvalho wrote:
 Can't I register multiple phones with the same user/password? That's
 what I pretend to do, not ring groups...

 Ricardo,

 Any particular reason for not using ring groups?

 No, you cannot register multiple phones with the same user/password.

 Just curious: can I register multiple phones with one user name but
 different passwords?

 no.
 ___

Which is relevant for asterisk (like any other client/server based
architecture), is the session.

Your phone (hard||soft) is the client.
Your PBX asterix is the server.

Your session is defined by your agent confiuration (and  configuration
data is sent in SIP protocol over TCP/IP suite protocol) .

But first there is a connection.(tcp/ip)

And on the same IP/PORT there is only one connection. If you change
username/password this is still one connection and the same connection.

username password are mostly used to authenticate and not to connect.

cheers

Bayrouni

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[asterisk-users] this i a test

2007-02-28 Thread Bayrouni

Sorry for disturbing, but I sent some messages today and I am not seeing
them on this list.
Can sombody tell me, in case this message appear on the list.

Thank you
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Re: [asterisk-users] Run-away Asterisk

2007-02-28 Thread Tzafrir Cohen

On Wed, Feb 28, 2007 at 10:56:14AM -0800, Yuan LIU wrote:
 After testing some AGI's, I noticed several extra Asterisk processes.  

An agi script is run by the same user running asterisk, but is not
asterisk: it is a different program. What is the command name on those
scripts?

 They 
 are not zombies, but can't be killed by safe_asterisk.  

safe_asterisk attempts (poorly) to guard asterisk. Not really to guard
all of its child processes.

 Nor will they die 
 when CLI issues stop now.  Then I read that each AGI spawns a separate 
 Asterisk process.  

Huh? AGI? FastAGI? 

 But all my AGI calls have apparently completed 
 successfully.  So there should be no reason for them to hang there.
 
 Several questions:
 
 1) Under what conditions will an AGI hang a process? (My test scripts are 
 pretty simple, almost directly derived from agi-test.agi.)

An AGI may be an arbitrary subprocess. This subprocess can do basically
everything. If it really wants to, (or if it misbehaves in the right
way) it won't die.

 
 2) How to detect run-away processes under 2.4 kernels?  In this kernel, 
 each thread clusters process space and it's very difficult to distinguish 
 them without killing the main process.

hmm, please attach the output of:

ps auxww | grep asterisk

 
 3) Any practical way to detect them from inside Asterisk - e.g., do some 
 check after each AGI call?  All my AGISTATUS reports success.  I could use 
 System() but isn't that cumbersome?

Write/use better code, I guess.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk

2007-02-28 Thread Webster, Andrew

Outbound calls on my Telus PRI aren't taking the Name portion of the
callerID. I've looked at the logs, and it is being set (see below), but
the PRI debug output doesn't show the name being sent anywhere. As a
result, received calls always display from Unknown (or just the number).
Is there some config that I've missed somewhere?

I'm running NI-1 (Telus says NI-2 doesn't support the name feature, so
they've changed my link type).
Version: Asterisk 1.2.14 svn rev 48468


Asterisk Log:
Executing Set(SIP/304-091aafb8, CALLERID(all)=Andrewnn) in
new stack
(I've replaced the digits with n).

PRI debug shows:
 Protocol Discriminator: Q.931 (8) len=42
 Call Ref: len= 2 (reference 4/0x4) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a2]
 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
 Ext: 1 User information layer 1: u-Law (34)
 [18 03 a9 83 81]
 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
 ChanSel: Reserved
 Ext: 1 Coding: 0 Number Specified Channel Type: 3
 Ext: 1 Channel: 1 ]
 [6c 0c 21 80 nn nn nn nn nn nn nn nn nn nn]
 Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 Presentation: Presentation permitted, user number not screened (0)
'nn' ]

From zapata.conf:
callerid=asreceived

;Sangoma A101 port 1 [slot:12 bus:0 span: 1]
switchtype=ni1
context=from-zaptel
overlapdial=yes
facilityenable=yes
group=0
signalling=pri_cpe
channel = 1-23

Thanks!
--
Andrew

 
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[asterisk-users] Newbie Planning Help

2007-02-28 Thread Alan Chandler

Excuse the ASCII diagramme - you will need a fixed width font to 
understand it.

 -- --- --- -
 | A  | == | NAT | === === | NAT | == | B |
 -- ---|   |--- -
  ---
  |  The Internet   |
  ---
| WAN interface (82.44.22.127)
  -
  | S   (NAT) |
  - 
   | LAN interface (192.168.0.20)
  =
  | 192.168.0.0/24 range  |
-- ---
| C  | | D   |
-- ---
I am at home on machine D (and with wife on machine C), with some family 
at machines A and B.

I am trying to setup an arrangement whereby clients on machines A, B, C 
and D can talk to each other on Softphones. A,B,C are are all Windows 
XP machines, machines D and S are linux.  This has to include A talking 
to B and ultimately conference calls with potentially all parties.

Machine S is my firewall/router providing NAT services to clients C and 
D (based soley on my own IPTABLES script) but is ALSO the machine I 
plan to put Asterisk on (it can therefore bind to two interfaces, with 
separate configurations for each if I so desire). If appropriate, I 
could install a STUN server on S.  I would prefer if media traffic 
between A and B avoids using my WAN interface pipe but if that is 
unavoidable, so be it.

I could use SIP or IAX softphones in this setup as long as it is no more 
complicated that telling A and B what to download and giving them 
simple setup instructions.  They could probably adjust their NAT 
routers to forward particular ports to them, but its not certain (A 
shares a flat with others).

I have a slight preference for SIP as it means I could potentially 
replace machines A,B and C with hardware devices in the future.

I have been round and round in circles reading the documentation but I 
am not sure I understand

a) to what extent Asterisk can manage everything necessary to allow 
machines A and B to communicate if they were SIP phones.  Is it 
possible to go for a setup with the firewalls/NAT devices as shown

b) if I go with IAX softphones, does communication between A and B have 
to go through S, or can Asterisk hand-off the IAX conversation so 
that A and B talk directly.

c) the example documentation shows seperate entries in iax.conf for 
incoming and outgoing calls.  In my case (assuming IAX softphones) 
would I just have entries for A and B of type friend?

Can someone give me some advice about how to proceed.

Thanks


-- 
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [asterisk-users] Changing from email address for vociemail.conf

2007-02-28 Thread Dovid B

I tried that as well and I get the same problem. Can it be an issue with 
sendmail ?


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

Sent: Wednesday, February 28, 2007 7:37 PM
Subject: Re: [asterisk-users] Changing from email address for vociemail.conf



-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

; Who the e-mail notification should appear to come from
;[EMAIL PROTECTED]


Dovid B wrote:

Hi List,
I put this in to my voicemail.conf as per the wikki and the users are
still getting the emails from the root account. Any ideas on what it can
be ?
I have: mailcmd=/usr/sbin/sendmail -v -t -f [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]





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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFF5b3ARhLSniguQyERAtrfAJ9XTsPRtgk/yxV/NivK36YgHvu7mQCdGMuU
awieqQ/FhGDyBZ4aAKjKioc=
=IIii
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Re: [asterisk-users] this i a test

2007-02-28 Thread Rodrigo Gonzalez


Bayrouni wrote:

Sorry for disturbing, but I sent some messages today and I am not seeing
them on this list.
Can sombody tell me, in case this message appear on the list.

Thank you
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Yes, here

Go to http://lists.digium.com/mailman/listinfo/asterisk-users

Login and check that you have Receive your own posts to the list? in yes 
if you want to receive your own emails

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[asterisk-users] 1.4 lost internet internal phones loose registration

2007-02-28 Thread Jerry Geis


I am running asterisk 1.4. I have 2 NICS in the my server.
Over the last couple days I have lost internet connection a couple times 
(lets not go there)...


Anyway everytime I loose internet my internal phones loose registration.

The phones are not using DNS they are coded to the servers IP.
The DHCP server is on the same local subnet so that is no issue.

Why would asterisk not be re-newing the registration to these local phones?

Jerry
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[asterisk-users] Send DTMF's before the call is answered

2007-02-28 Thread Álvaro Palma


Is there a way to send DTMF's to a channel before the call is answered?

For example, send DTMF's to a SIP channel after the 180 Ringing or 183 
Session Progress have been received from it, but before the 200 OK, or 
in the E1 side, after the Q931_ALERTING is received, but before the 
Q931_CONNECT. If I use Dial(SIP/,D(my_dtmfs)), it will wait until 
SIP/ have answered to send the tones, but I need to do it before that.


Thanks a lot for your help.

--
Attn.
Alvaro Palma
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Re: [asterisk-users] Newbie Planning Help

2007-02-28 Thread mail-lists



a) to what extent Asterisk can manage everything necessary to allow 
machines A and B to communicate if they were SIP phones.  Is it 
possible to go for a setup with the firewalls/NAT devices as shown
  
If the asterisk machine isn't NATed you shouldn't have a problem at all. 
If you're using SIP clients just make sure nat=yes

is set in each of the client definitions in sip.conf

b) if I go with IAX softphones, does communication between A and B have 
to go through S, or can Asterisk hand-off the IAX conversation so 
that A and B talk directly.
  
I'm not sure in this case since both clients are going to be NATed. I'm 
pretty sure that this wouldn't work with SIP clients.
Since IAX has less problems with NAT traversal it might work fine - try 
setting canreinvite=yes in your iax.conf and monitor

rtp traffic at the asterisk CLI
c) the example documentation shows seperate entries in iax.conf for 
incoming and outgoing calls.  In my case (assuming IAX softphones) 
would I just have entries for A and B of type friend?


Can someone give me some advice about how to proceed.
  

type=friend works for me...


If you decide to use iax check out moziax - a firefox plugin iax client 
that's simple to set up.

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Re: [asterisk-users] running asterisk through cellphone

2007-02-28 Thread Michael Kamleitner

On 2/28/07, Dovid B [EMAIL PROTECTED] wrote:

Cant take the credit. I didnt create it. as far as a phone you can go
with 2 things. either use chan_cellphone and use bluetooth or you can go
with a cell phone dock (as some one mentioned earlier). if you are using the
cellular docking station that you dont need to worry about chan_cellphone.
in regards to your other question once the phone is set up it will act like
a regular line. once asterisk is connected to it it's just a matter of
setting up the dial plan to do what you want it to do.

sounds great...I think I'll try it via bluetooth first...

or if anyone could recommend me a particular phone to try this with...?

thx everybody, can't wait to get this going :)

michael

- Original Message -

*From:* Michael Kamleitner [EMAIL PROTECTED]
*To:* Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
*Sent:* Tuesday, February 27, 2007 9:21 PM
*Subject:* Re: [asterisk-users] running asterisk through cellphone

hi dovid,

thx for replying, as I can see the chan_cellphone patch was done by you,
great! looks like this is exactly what I want. my goal is to connect a
normal consumer cellphone to the asterisk-server, allowing anyone else to
phone-in from their regular phone.

it would be even better if I could use this setup to emulate extension -
so lets assume my cellphone-number is 004369912345678, than I would like to
have 3 separate extensions at 004369912345678-01, 004369912345678-02 and
004369912345678-03. is this possible?

as I'm going to buy a separate phone for this task, can anyone recommend
certain models (besides the RIM blackberry mentioned in the docs)?

greetings,
michael

On 2/27/07, Dovid B [EMAIL PROTECTED] wrote:

What is the cellular connection for ? Are you using this for inbound or
the clients will call in in from thier cell phones ? If you need incoming
(and or ourgoing) lines you can get one from an ITSP. If you want to use
your cell phone you can use chan_cellphone. In order to use it you will need
to install the patch. For more information have at look at this:
http://bugs.digium.com/view.php?id=8919

http://bugs.digium.com/view.php?id=8919http://bugs.digium.com/view.php?id=8919http://bugs.digium.com/view.php?id=8919

- Original Message -
*From:* Michael Kamleitner [EMAIL PROTECTED]
*To:* asterisk-users@lists.digium.com
*Sent:* Tuesday, February 27, 2007 6:54 PM
*Subject:* [asterisk-users] running asterisk through cellphone

hi everybody,

I'm currently planning a small-sized web-applicaiton allowing users to
call-in via phone. the phonecalls should be recorded and processed further
by some custom scripts - sounds like asterisk is a perfect match for this
app.

however, during prototyping I have no ISDN-connection whatsoever
available, so I was asking myself if it's possible to connect a cellphone
via data-cable (or bluetooth?) and use this as the single line to call-in.
searching the asterisk-forums I found mentions of chan_cellphone, which
is probably a patch for exactly this kind of usage, right?

I'ld be thankful if you could just point me to the right direction (I'm
quite new to asterisk). thx in advance!

michael

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--
10 Jahre The Gap
Party am 15.3.2007 - www.tengap.at

Mag. Michael Kamleitner
-
[EMAIL PROTECTED]
+43 699 11607923
https://www.xing.com/profile/Michael_Kamleitner
-
m-otion GmbH
Favoritenstr 4-6/III, 1040 Wien
+43 1 205705 / 21 (Fax 99)
-
www.m-otion.com

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--
10 Jahre The Gap
Party am 15.3.2007 - www.tengap.at

Mag. Michael Kamleitner
-
[EMAIL PROTECTED]
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https://www.xing.com/profile/Michael_Kamleitner

Re: [asterisk-users] TE212P on FC6 - stack overflow?

2007-02-28 Thread Matthew Fredrickson

Try latest zaptel 1.2 from svn.  I made a fix that should reduce stack 
usage.


Matthew Fredrickson

On Feb 27, 2007, at 4:49 PM, Marco Parisotto wrote:



Hi Michelle,

actually, I didn't try it...
The server is a HP Proliant ML150T G3.
Currently I'm not in the condition to follow your suggestion, but I 
hope in the near future to be able to give you a feedback.


Thanks!
Marco

 Have you tried starting Linux with irqpoll / noapic?  Sounds like a 
BIOS

 bug..

 MD

  _

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Marco
Parisotto
Sent: Tuesday, February 27, 2007 3:27 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] TE212P on FC6 - stack overflow?


Hi all
did anyone of you experience an error like do_irq: stack overflow in
configuring a TE212P on Fedora core 6? The server immediately hangs, I
 don't
know if this can be related to hardware configuration or kernel
incompatibility... This problem arises when I try to configure the 
channels
with the usual command ztcfg and it is strictly related to the 
presence of

the echo canceller
 onboard.

Thanks a lot
Marco

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[asterisk-users] Occasional SMS problem

2007-02-28 Thread Arik Raffael Funke


Hi,

I am using asterisk's SMS functionality for sending messages. Most of 
the time it works without problems (as in situation 1) but sometimes 
something seems to go wrong with the transmission (as in situation 2). I 
am using Deutsche Telekom, Germany's main TELCO, so I suppose the 
problem must be on my end. Can anybody tell me what is going on or how I 
could narrow down the problem?


Cheers,
Arik


Situation 1
===
-- Attempting call on mISDN/g:extern/0193010 for application SMS(0) 
(Retry 1)

funke*CLI
funke*CLI
Channel mISDN/2-u11 was answered.
Launching SMS(0) on mISDN/2-u11
-- SMS RX 93 00 6D
-- SMS TX 91 1C 01 0B 0C 81 70 21 41 43 58 03 00 F1 11 C4 74 79 0E 
4A CF E9 A0 72 DA 0D A2 96 E7 74...

-- SMS RX 95 09 01 00 70 20 82 12 55 70 40 38
-- SMS TX 94 00 6C
Feb 28 21:55:09 NOTICE[1963]: pbx_spool.c:279 attempt_thread: Call 
completed to mISDN/g:extern/0193010



Situation 2
===
-- Attempting call on mISDN/g:extern/0193010 for application SMS(0) 
(Retry 1)

Channel mISDN/2-u10 was answered.
Launching SMS(0) on mISDN/2-u10
-- SMS RX 93 00 6D
-- SMS TX 91 1C 01 0A 0C 81 70 21 41 43 58 03 00 F1 11 C4 74 79 0E 
4A CF E9 A0 72 DA 0D A2 96 E7 74...

-- SMS RX 95 09 01 00 70 20 82 12 25 84 40 54
-- SMS TX 94 00 6C
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
Feb 28 21:53:28 NOTICE[1872]: pbx_spool.c:279 attempt_thread: Call 
completed to mISDN/g:extern/0193010


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Re: [asterisk-users] Newbie Planning Help

2007-02-28 Thread Andrew Kohlsmith

On Wednesday 28 February 2007 3:45 pm, Alan Chandler wrote:
 I am trying to setup an arrangement whereby clients on machines A, B, C
 and D can talk to each other on Softphones. A,B,C are are all Windows
 XP machines, machines D and S are linux.  This has to include A talking
 to B and ultimately conference calls with potentially all parties.

Personally I make my Asterisk box the firewall.  It eliminates all NAT 
troubles.  :-)

If that's not your style, I'd use IAX over SIP, as it only requires a 
port-forward to D on D's NAT box.  SIP you may be able to get work with port 
forwarding 5060 and 1-2 (all udp) over to D, but I'm not sure... 
Naturally, nat=yes and canreinvite=no should be set all around.

-A.
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Re: [asterisk-users] 1.4 lost internet internal phones loose registration

Asterisk gets very upset when DNS is down.  You might want to confirm 
that /etc/hosts has entries for ALL interfaces in that system.  That 
should cause the system to not issue a DNS request to resolve local IPs.


Jerry Geis wrote:

I am running asterisk 1.4. I have 2 NICS in the my server.
Over the last couple days I have lost internet connection a couple times 
(lets not go there)...


Anyway everytime I loose internet my internal phones loose registration.

The phones are not using DNS they are coded to the servers IP.
The DHCP server is on the same local subnet so that is no issue.

Why would asterisk not be re-newing the registration to these local phones?

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[asterisk-users] No Caller ID Name PRI NI2

2007-02-28 Thread foucaulom


I there,

I have some trouble to do working caller id name for outgoing calls on 
the PRI we just installed. Caller id name work on incoming calls only.

Caller id number work on incoming and outgoing calls.


The provider, Goup Telecom, said that's in what i'm sending. They said 
that I send the cid name in ascii code and to do it working, I need to 
send it in hex.


So I take some traces but i'm unable to figure where is the problem.

What I see In case that work: incoming call:
 [1c 1c 9f 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f 52 
54 49 4e 20 46 41 58]
 Facility (len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16, 
0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0E, 'INFOFORTIN FAX' ]

PROTOCOL 1F

What I see in case that doesn't work: outgoing call:

[28 05 b1 69 6e 66 6f]
Display (len= 5) Charset: 31 [ info ]



completes traces:

working:
 [ 02 01 da d6 08 02 02 34 05 04 03 80 90 a2 18 03 a9 83 81 1c 1c 9f 
8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f 52 54 49 4e 20 
46 41 58 1e 02 82 83 6c 0c 21 83 38 31 39 37 38 30 31 32 37 33 70 0b a1 
38 31 39 33 34 30 30 39 37 37 ]

 Informational frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
 N(S): 109   0: 0
 N(R): 107   P: 0
 76 bytes of data
-- ACKing all packets from 106 to (but not including) 107
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
 Protocol Discriminator: Q.931 (8)  len=76
 Call Ref: len= 2 (reference 564/0x234) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a2]
 Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)

  Ext: 1  User information layer 1: u-Law (34)
 [18 03 a9 83 81]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  
Exclusive  Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
   Ext: 1  Channel: 1 ]
 [1c 1c 9f 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f 52 
54 49 4e 20 46 41 58]
 Facility (len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16, 
0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0E, 'INFOFORTIN FAX' ]

PROTOCOL 1F
8B 0001 00 (CONTEXT SPECIFIC [11])
A1 0016 (CONTEXT SPECIFIC [1])
 02 0001 01 (INTEGER: 1)
 02 0001 00 (INTEGER: 0)
 80 000E 49 4E 46 4F 46 4F 52 54 49 4E 20 46 41 58 (CONTEXT SPECIFIC [0])
 [1e 02 82 83]
 Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard 
(0)  0: 0  Location: Public network serving the local user (2)
   Ext: 1  Progress Description: Calling 
equipment is non-ISDN. (3) ]

 [6c 0c 21 83 38 31 39 37 38 30 31 32 37 33]
 Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation allowed of 
network provided number (3)  '8197801273' ]

 [70 0b a1 38 31 39 33 34 30 30 39 37 37]
 Called Number (len=13) [ Ext: 1  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '8193400977' ]

-- Making new call for cr 564
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 28 (cs0, Facility)
Q.932 Interpretation component is not handled
Handle Q.932 ROSE Invoke component
 [ Handling operation 0 ]
 Handle Name display operation
   Received caller name 'INFOFORTIN FAX'
-- Processing IE 30 (cs0, Progress Indicator)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
q931.c:3294 q931_receive: call 564 on channel 1 enters state 6 (Call Present)
Sending Receiver Ready (110)

[ 02 01 01 dc ]
Supervisory frame:
SAPI: 00  C/R: 1 EA: 0
 TEI: 000EA: 1
Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
N(R): 110 P/F: 0
0 bytes of data

-- Restarting T203 counter
-- Restarting T203 counter
q931.c:2570 q931_call_proceeding: call 564 on channel 1 enters state 9 
(Incoming Call Proceeding)

[ 00 01 d6 dc 08 02 82 34 02 18 03 a9 83 81 ]
Informational frame:
SAPI: 00  C/R: 0 EA: 0
 TEI: 000EA: 1
N(S): 107   0: 0
N(R): 110   P: 0
10 bytes of data

-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer

Protocol Discriminator: Q.931 (8)  len=10
Call Ref: len= 2 (reference 564/0x234) (Terminator)
Message type: CALL PROCEEDING (2)
[18 03 a9 83 81]




don't working:

-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer

Protocol Discriminator: Q.931 (8)  len=54
Call Ref: len= 2 (reference 13/0xD) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
 Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)

 Ext: 1  User information

[asterisk-users] Newbie extensions.conf question

2007-02-28 Thread Chris Griffin

I've installed Sven Slezak's Notify module. He gives the follow as an  
example line to put into extensions.conf


exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/ 
sunnybook)


I understand what is going on with this line but I don't know where  
in the extensions.conf file to put it?


Thanks,
Chris Griffin
[EMAIL PROTECTED]



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[asterisk-users] extensions.conf sccp.conf howto call external number

2007-02-28 Thread Daniel Schlager


Hello @List,

i'm using a Cisco 7970 / 7914 phone with Asterisk 1.4  sccp

part of my sccp.conf

type= 7914 (Cisco 7970 with 7914 phone extension)
autologin   = 117
description = Test
speeddial   = 10,Test (10),[EMAIL PROTECTED]

my speeddial line is for internal call

howto call a external number eg. +xx467584933 ??
i want to press a button and call directly out

thank you

greetings,
Daniel
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[asterisk-users] OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED

2007-02-28 Thread Kristian Kielhofner


Hey everyone,

 I came across a situation where I needed to use CDP to advertise a
voice vlan to Polycom/Cisco (and other CDP capable phones) without a
Cisco switch.

 Sure I have Cisco switches in places but I like my Polycoms to work
out of the box and it isn't always practical to purchase a Cisco
switch for every location.

cdp-tools homepage:

http://gpl.internetconnection.net/

 So I found cdp-tools to try to advertise the voice vlan using CDP.
On Cisco switches this is used with switchport voice vlan  where
xxx is the vlan.  But what if you don't have a Cisco switch or your
switch doesn't support CDP?

 I grabbed cdp-tools and started playing around.  You will need to
install libpcap (and headers if you are using package management) and
libnet (preferably 1.1 and headers).

 I got to work on cdp-tools and found that there were several problems:

- It only advertised CDPv1 (current is CDPv2)
- It didn't support Voice VLANs
- It didn't properly support native VLANs
- It didn't properly support advertising duplex

 I was fortunate enough to have access to some real Cisco switches.
I was able to grab the CDP advertisements (after defining a voice
vlan) and decode them using ethereal (wireshark).

 Using my handy capture of a correct CDP advert from a 2960G (CDP
v2), I started working on the cdp-tools source.  I was able to correct
all of the problems above.  I added a new argument (-V) to specify a
voice vlan to advertise.

 After some quick testing, I was able to trick a Polycom into
thinking that my laptop was a CDPv2 capable switch and the Polycom (IP
601) successfully discovered the correct VLAN when connected to my
laptop with a crossover cable (I had already used vconfig to add the
voice vlan to the laptop).

 A quick howto:

- Download and extract cdp-tools:
http://gpl.internetconnection.net/files/cdp-tools.tar.gz

- Apply my patch
http://www.krisk.org/asterisk/cdp-tools.patch

- Compile cdp-tools

- Run cdp-tools (as root):
sudo ./cdp-send -c l2sw -d -L 1 -V 20 -m i586 -n pbx -s FakeSwitch
-t 10 -P full eth0

Explanation:
-c (advertise as layer 2 switch)
-d (turn debugging on)
-L (Native VLAN)
-V (Voice VLAN)
-m (machine architecture)
-n (machine name)
-s (machine software)
-t (wait time between broadcasts)
-P (advertise duplex)

and eth0 is the interface to broadcast on...

 You should now be advertising VLAN 20 as your voice vlan and VLAN
1 as your default VLAN.  Your machine will also be visible from any
CDP v2 capable connected devices.  Try sh cdp neigh from any Cisco
switches on your network.  You should see your Linux machine!


 TO-DO:

- CDPv2 support is a very loose term...  It is NOWHERE near complete
and it appears to be the bare minimum to work with a Polycom phone.

- The variable names in my patch suck!

 I would appreciate some testing before I send this patch upstream.
Let me know how it works out (if at all).

Thanks!

--
Kristian Kielhofner
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RE: [asterisk-users] No Caller ID Name PRI NI2

2007-02-28 Thread Webster, Andrew


Hi,

I'm having a similar problem, but the name isn't even appearing in debug
output of PRI (see my other post about this).  My PRI is with Telus, and
they told me that NI-2 doesn't support CallerID name function, only
NI-1.


Andrew


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
 Sent: Wednesday, February 28, 2007 17:03
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] No Caller ID Name PRI NI2
 
 I there,
 
 I have some trouble to do working caller id name for outgoing calls on
 the PRI we just installed. Caller id name work on incoming calls only.
 Caller id number work on incoming and outgoing calls.
 
 
 The provider, Goup Telecom, said that's in what i'm sending. They said
 that I send the cid name in ascii code and to do it working, I need to
 send it in hex.
 
 So I take some traces but i'm unable to figure where is the problem.
 
 What I see In case that work: incoming call:
  [1c 1c 9f 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f
52
 54 49 4e 20 46 41 58]
  Facility (len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16,
 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0E, 'INFOFORTIN FAX' ]
 PROTOCOL 1F
 
 What I see in case that doesn't work: outgoing call:
  [28 05 b1 69 6e 66 6f]
  Display (len= 5) Charset: 31 [ info ]
 
 
 completes traces:
 
 working:
  [ 02 01 da d6 08 02 02 34 05 04 03 80 90 a2 18 03 a9 83 81 1c 1c 9f
 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f 52 54 49 4e
20
 46 41 58 1e 02 82 83 6c 0c 21 83 38 31 39 37 38 30 31 32 37 33 70 0b
a1
 38 31 39 33 34 30 30 39 37 37 ]
  Informational frame:
  SAPI: 00  C/R: 1 EA: 0
   TEI: 000EA: 1
  N(S): 109   0: 0
  N(R): 107   P: 0
  76 bytes of data
 -- ACKing all packets from 106 to (but not including) 107
 -- Since there was nothing left, stopping T200 counter
 -- Stopping T203 counter since we got an ACK
 -- Nothing left, starting T203 counter
  Protocol Discriminator: Q.931 (8)  len=76
  Call Ref: len= 2 (reference 564/0x234) (Originator)
  Message type: SETUP (5)
  [04 03 80 90 a2]
  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
 capability: Speech (0)
   Ext: 1  Trans mode/rate: 64kbps,
 circuit-mode (16)
   Ext: 1  User information layer 1: u-Law
 (34)
  [18 03 a9 83 81]
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
 Exclusive  Dchan: 0
 ChanSel: Reserved
Ext: 1  Coding: 0  Number Specified  Channel
Type:
 3
Ext: 1  Channel: 1 ]
  [1c 1c 9f 8b 01 00 a1 16 02 01 01 02 01 00 80 0e 49 4e 46 4f 46 4f
52
 54 49 4e 20 46 41 58]
  Facility (len=30, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x16,
 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0E, 'INFOFORTIN FAX' ]
 PROTOCOL 1F
 8B 0001 00 (CONTEXT SPECIFIC [11])
 A1 0016 (CONTEXT SPECIFIC [1])
   02 0001 01 (INTEGER: 1)
   02 0001 00 (INTEGER: 0)
   80 000E 49 4E 46 4F 46 4F 52 54 49 4E 20 46 41 58 (CONTEXT SPECIFIC
[0])
  [1e 02 82 83]
  Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
 (0)  0: 0  Location: Public network serving the local user (2)
Ext: 1  Progress Description: Calling
 equipment is non-ISDN. (3) ]
  [6c 0c 21 83 38 31 39 37 38 30 31 32 37 33]
  Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation allowed of
 network provided number (3)  '8197801273' ]
  [70 0b a1 38 31 39 33 34 30 30 39 37 37]
  Called Number (len=13) [ Ext: 1  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '8193400977' ]
 -- Making new call for cr 564
 -- Processing Q.931 Call Setup
 -- Processing IE 4 (cs0, Bearer Capability)
 -- Processing IE 24 (cs0, Channel Identification)
 -- Processing IE 28 (cs0, Facility)
 Q.932 Interpretation component is not handled
 Handle Q.932 ROSE Invoke component
   [ Handling operation 0 ]
   Handle Name display operation
 Received caller name 'INFOFORTIN FAX'
 -- Processing IE 30 (cs0, Progress Indicator)
 -- Processing IE 108 (cs0, Calling Party Number)
 -- Processing IE 112 (cs0, Called Party Number)
 q931.c:3294 q931_receive: call 564 on channel 1 enters state 6 (Call
 Present)
 Sending Receiver Ready (110)
  [ 02 01 01 dc ]
  Supervisory frame:
  SAPI: 00  C/R: 1 EA: 0
   TEI: 000EA: 1
  Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
  N(R): 110 P/F: 0
  0 bytes of data
 -- Restarting T203 counter
 -- Restarting T203 counter
 q931.c:2570 q931_call_proceeding: call 564 on channel 1 enters state 9
 (Incoming Call Proceeding)
  [ 00 01 d6 dc 08 02 82 34 02 18 03 a9 83 81 ]
  Informational frame:
  SAPI: 00  C/R: 0 EA: 0
   TEI: 000EA: 1
  N(S): 107   0: 0
  N(R): 110   P: 0
  10 bytes of data
 -- Restarting T203 counter
 Stopping T_203 timer
 Starting T_200

Re: [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk


nice one.. we have rogers and primus.. ni'2 and same..

let me know if this ni2 and ni1 thing is crap or not

On 2/28/07, Webster, Andrew [EMAIL PROTECTED] wrote:


 Outbound calls on my Telus PRI aren't taking the Name portion of the
callerID. I've looked at the logs, and it is being set (see below), but the
PRI debug output doesn't show the name being sent anywhere. As a result,
received calls always display from Unknown (or just the number).
Is there some config that I've missed somewhere?

I'm running NI-1 (Telus says NI-2 doesn't support the name feature, so
they've changed my link type).
Version: Asterisk 1.2.14 svn rev 48468


Asterisk Log:
Executing Set(SIP/304-091aafb8, CALLERID(all)=Andrewnn) in
new stack
(I've replaced the digits with n).

PRI debug shows:
 Protocol Discriminator: Q.931 (8) len=42
 Call Ref: len= 2 (reference 4/0x4) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a2]
 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
 Ext: 1 User information layer 1: u-Law (34)
 [18 03 a9 83 81]
 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
 ChanSel: Reserved
 Ext: 1 Coding: 0 Number Specified Channel Type: 3
 Ext: 1 Channel: 1 ]
 [6c 0c 21 80 nn nn nn nn nn nn nn nn nn nn]
 Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 Presentation: Presentation permitted, user number not screened (0)
'nn' ]

From zapata.conf:
callerid=asreceived

;Sangoma A101 port 1 [slot:12 bus:0 span: 1]
switchtype=ni1
context=from-zaptel
overlapdial=yes
facilityenable=yes
group=0
signalling=pri_cpe
channel = 1-23

Thanks!
--
Andrew



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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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[asterisk-users] read write or only read fields in cdr?

2007-02-28 Thread Bayrouni

Hello,


I created a new field named pre_dst of type varchar(80) exactly like dst
 field in cdr table.

In the dialplan I put:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})

and when I call, all goes fine except that pre_dst has always NULL value
in cdr.

Do you know why?
Is something wrong I did?


I know that original fields in cdr are only readable, but in this cas
pre_dst is one I created myself !!!

Thank you.



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[asterisk-users] Paid support offered


We have decided to allow our tech's to do support for non-clients of
voicemeup.com

You can head to http://support.voicemeup.com/ and one will be in touch 8 to
6pm business hours.

3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc.



--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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[asterisk-users] Newbie Planning Help

2007-02-28 Thread Gleim, Jason

Answers in-line...

Hope this helps!
Jason

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alan
Chandler
Sent: Wednesday, February 28, 2007 3:46 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie Planning Help

snip
-- ---

a) to what extent Asterisk can manage everything necessary to allow 
machines A and B to communicate if they were SIP phones.  Is it 
possible to go for a setup with the firewalls/NAT devices as shown

- Asterisk can register and manage both A  B even though they
are behind NAT devices. NAT=yes is required, of course, for Asterisk and
the endpoint to properly communicate. You probably know but just in
case, SIP endpoints maintain a signaling channel through port 5060. When
a call comes in, they open a RTP media stream somewhere between port
1 and port 2. NAT can sometimes mess this up and it usually
shows itself as one-way audio. IAX endpoints send signaling and media
over the same port so there is less risk in NAT problems.

b) if I go with IAX softphones, does communication between A and B have 
to go through S, or can Asterisk hand-off the IAX conversation so 
that A and B talk directly.

- I do not believe IAX allows for a hand-off between the two
endpoints. Most people don't want the hand-off anyway as it prevents the
parties from using in-call feature codes. This is why most everyone sets
canreinvite=no for SIP endpoints.

c) the example documentation shows seperate entries in iax.conf for 
incoming and outgoing calls.  In my case (assuming IAX softphones) 
would I just have entries for A and B of type friend?

- yes. 'friend' is you friend for IAX softphones!

Can someone give me some advice about how to proceed.

Thanks


-- 
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [asterisk-users] Newbie Planning Help

2007-02-28 Thread Alan Chandler

On Wednesday 28 February 2007 21:26, Andrew Kohlsmith wrote:
 On Wednesday 28 February 2007 3:45 pm, Alan Chandler wrote:
  I am trying to setup an arrangement whereby clients on machines A,
  B, C and D can talk to each other on Softphones. A,B,C are are all
  Windows XP machines, machines D and S are linux.  This has to
  include A talking to B and ultimately conference calls with
  potentially all parties.

 Personally I make my Asterisk box the firewall.  It eliminates all
 NAT troubles.  :-)

Yes thats what I meant.  My box S is the firewall and * will run on it.  
BUT, both A and B will have NAT firewall/routers outside of them AND 
somehow C and D will need to go through the S (does the traffic go 
round the outside of * or through the middle of it?


 If that's not your style, I'd use IAX over SIP, as it only requires a
 port-forward to D on D's NAT box.  SIP you may be able to get work
 with port forwarding 5060 and 1-2 (all udp) over to D, but
I am not sure I am following.  Why is D different from C? if I port 
forward everything to D how does C get into the conversation

 I'm not sure... Naturally, nat=yes and canreinvite=no should be set
 all around.

Why? and doesn't the canreinvite=no mean all the traffic from A to B 
goes through S, something I would prefer to avoid.


-- 
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [asterisk-users] Newbie Planning Help

2007-02-28 Thread Alan Chandler

On Wednesday 28 February 2007 21:08, mail-lists wrote:
  a) to what extent Asterisk can manage everything necessary to allow
  machines A and B to communicate if they were SIP phones.  Is it
  possible to go for a setup with the firewalls/NAT devices as shown

 If the asterisk machine isn't NATed you shouldn't have a problem at
 all. If you're using SIP clients just make sure nat=yes
 is set in each of the client definitions in sip.conf

  b) if I go with IAX softphones, does communication between A and B
  have to go through S, or can Asterisk hand-off the IAX
  conversation so that A and B talk directly.

 I'm not sure in this case since both clients are going to be NATed.
 I'm pretty sure that this wouldn't work with SIP clients.

Now you have confused me.  In the answer to a) you say that for each SIP 
client I say nat=yes and it will work, yet here you say this wouldn't 
work if both clients are going to be SIP.

 Since IAX has less problems with NAT traversal it might work fine -
 try setting canreinvite=yes in your iax.conf and monitor
 rtp traffic at the asterisk CLI

You have confused me again.  I thought the point of IAX is that there 
isn't any separate RTP traffic.


  c) the example documentation shows seperate entries in iax.conf for
  incoming and outgoing calls.  In my case (assuming IAX softphones)
  would I just have entries for A and B of type friend?
 
  Can someone give me some advice about how to proceed.

 type=friend works for me...

I am not sure where all that leaves me.  Should I use SIP everywhere or 
IAX everywhere
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Re: [asterisk-users] read write or only read fields in cdr?

2007-02-28 Thread Edgar Luna

Hi,

On Wed, 2007-02-28 at 23:43 +0100, Bayrouni wrote:
 Hello,

 In the dialplan I put:
 exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
 
 and when I call, all goes fine except that pre_dst has always NULL value
 in cdr.
 
 Do you know why?
 Is something wrong I did?

As far as I know, custom fields doesn't work with any database backend,
only with CSV. There is an addon in the bug tracker but seems that it
isn't finished.


Regards.
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Re: [asterisk-users] read write or only read fields in cdr?


try not using dst.. maybe its a regex on te fieldname that matches for
reserved keywords..

try pre_dest instead

On 2/28/07, Bayrouni [EMAIL PROTECTED] wrote:


Hello,


I created a new field named pre_dst of type varchar(80) exactly like dst
field in cdr table.

In the dialplan I put:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})

and when I call, all goes fine except that pre_dst has always NULL value
in cdr.

Do you know why?
Is something wrong I did?


I know that original fields in cdr are only readable, but in this cas
pre_dst is one I created myself !!!

Thank you.



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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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[asterisk-users] AEL Blacklist question

2007-02-28 Thread Neil Cherry


Does the ${BLACKLIST()} function allow for values other than 1 to be
returned and if so how can I use that is the AEL? Can I use the
function in a switch statement?

--
Linux Home Automation Neil Cherry   [EMAIL PROTECTED]
http://www.linuxha.com/ Main site
http://linuxha.blogspot.com/My HA Blog
Author of:  Linux Smart Homes For Dummies
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Re: [asterisk-users] Newbie extensions.conf question


try putting near the exten = 1000,1,dial stuff

On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:


I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf

exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)

I understand what is going on with this line but I don't know where
in the extensions.conf file to put it?

Thanks,
Chris Griffin
[EMAIL PROTECTED]



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Mike
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Making it happen
1.877.807.VOIP (8647)
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Re: [asterisk-users] Run-away Asterisk