Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

[dave cantera]

jeronimo
there is no difference...

Jeronimo Romero wrote:

  Is there any technical difference between a T1 cable and a cat5e patch
cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards using
straight through CAT5e cables? If so, are they using all of the pins in
the cable?

Thanks in advance
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RE: [asterisk-users] Follow me on multiple numbers..

[Philippe Lindheimer]

I'm not sure what problem you are having, it should be fine on a proper freepbx 
install (and as best I know trixbox installs it properly although I don't use 
trixbox so can't say first hand). It should work fine, I know plenty of people 
who use it - I use it regularly. Feel free to try and catch me or someone else 
on the freepbx IRC to get help if you have it setup so you can take a look. 
(although I'll be gone all week cause of VON).
   
  philippel

  
From: Kevin Kiely <[EMAIL PROTECTED]>
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Date: Sun, 18 Mar 2007 11:33:07 -0400
Subject: RE: [asterisk-users] Follow me on multiple numbers..

  

v\:* {behavior:url(#default#VML);}  o\:* {behavior:url(#default#VML);}  
w\:* {behavior:url(#default#VML);}  .shape {behavior:url(#default#VML);}
I tried to look at the code in Trixbox but when the option ‘confirm’ is 
selected in the follow me properties screen, no code is generated and the call 
goes dead.  Is there a trick to get the code generated?
   
   
  
-
  
  From: Philippe Lindheimer [mailto:[EMAIL PROTECTED] 
Sent: Saturday, March 17, 2007 12:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Follow me on multiple numbers..

   
  On 3/16/07, Ritesh Agrawal <[EMAIL PROTECTED]> wrote:

> Hi Folks,
  >
  > I want to setup a follow me routine so that asterisk can call me on the
  > multiple numbers.
  > I tried some of the samples at voip-info but there is a problem with those
  > examples.
  >
  > I dont have coverage in my home area and my cell phone answering machine
  > picks up the phone right away so my home phone never rings.
  > I also want the caller to be able to leave a voicemail and the cell phone
  > answering machine messes it all up.
  > I have call screening setup so the call gets answered by the cell phone
  > answering machine and it never accepts the call.
  >
  > I would appreciate if someone can help me with the setup.
  >
  You can create a follow-me

 with 1.2 that requires you to confirm the call before
  answering the channel. If you need an example, go have a look at the code I
  generate in the dialplan in freepbx to do that exact thing when you choose 
call
  confirm. No need to go to 1.4 just for that.
  
  philippel



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

[Jeronimo Romero]

The revision on this card was new and my version of zaptel (1.2.11) did
not support my pri signaling for that card. I upgraded zaptel through
svn and all was good. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Wednesday, February 28, 2007 12:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] TE110P: Error ==> Asterisk died with code
1.

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?

-- 
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[asterisk-users] zttool always reports "OK" on TDM400P

[Yuan LIU]

Just noticed that no matter what the line condition is, zttool always 
reports "OK", so it's pretty useless. (In contrast, I'd get "Red alert" if I 
unplug the line connecting to an X100P.)


I'm using zaptel 1.2.15 on Linux 2.6.15-28 (also tested on 2.6.10).

Yuan Liu


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Re: [asterisk-users] camp on off-line phone

[Philipp Kempgen]

Jeronimo Romero wrote:

> It would be cool if you could add some kind of login script capability to 
> nodes in  sip.conf and iax.conf.
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
> Sent: Sunday, March 18, 2007 11:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] camp on off-line phone
> 
> Leif Neland wrote:
> 
>> When phone A registers, I want phone B to ring, when picked up, it should 
>> call phone A and connect the phones.
>>
>> Translated: When GF in Mexico powers up laptop where soft iax-phone 
>> registers automatically, I want to talk to her asap :-)
>>
>> How to?

Maybe manager events are generated for SIP registrations? Haven't
checked. So you could connect to the AMI, listen to the events
and do something as soon as the girlfriend registers.


Regards,
  Philipp

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Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

[John Novack]

Learn to search - this question has been asked and answered at least 100 
times

There are also all sorts of cable references on the Internet

Can you spell "google" ?


Jeronimo Romero wrote:

So a regular cross over cable wouldn't work?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Sunday, March 18, 2007 10:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

 Yes. At that point, you're looking for a T1 "cross-over".

 The pinout is as follows:

1

4

RX/Ring/- <-->TX/Ring/-
2

5

RX/Tip/+ <-->TX/Tip/+
4

1

TX/Ring/- <-->RX/Ring/-
5

2

TX/Tip/+ <-->RX/Tip/+
3

3

Shield/Return/Ground
6

6

Shield/Return/Ground

On 3/18/07, Jeronimo Romero <[EMAIL PROTECTED]> wrote:
  

I assume that I would need to cross these pins over if I were going


from
  

t1 card to t1 card. Is this correct?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Sent: Sunday, March 18, 2007 7:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

A common Cat5 straight through cable will work fine.

T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for
signals.

A T1 loopback plug would be wired 1 to 4 and 2 to 5.

They come in handy for testing T1 cards or for providing a hard loop
for the telco.

Tom

At 05:42 PM 3/18/2007, you wrote:


Is there any technical difference between a T1 cable and a cat5e
  

patch
  

cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards
  

using
  

straight through CAT5e cables? If so, are they using all of the pins
  

in
  

the cable?

Thanks in advance
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[asterisk-users] proposal: a new mailing list for asterisk 1.4, why not?

[Angel Heart]

Hi Giorgio,

I guess it will be more benefitial to all old version users to read some 
information regarding new version like * 1.4. In this way, they will be 
encourage, or probably have an idea whether to upgrade or not based on all  the 
concerns that was posted. I for one still using 1.2.13 but I love reading * 1.4 
concerns before leaping to 1.4.

Regards,

Angel 

dave cantera <[EMAIL PROTECTED]> wrote:here! here!  they are different 
beasts...
 
 Giorgio Incantalupo wrote: Hi all,   
 since Asterisk 1.4 seems to have too many differences from previous versions, 
wouldn't be nice to have a new mailing list?   
   
 Giorgio Incantalupo   
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Re: [asterisk-users] Problems with MFCR2 and Meridian

[Angel Heart]

Hi Artur,

Just follow the information Moises had recommended you and for sure this will 
work. The default configurations that was exampled in the document is just 
fined and suited with Nortel Meridian. Just be sure that your Nortel MFC Card 
is installed and working in good condition with up to date Nortel patches.

Regards,

Angel


Moises Silva <[EMAIL PROTECTED]> wrote: Arturo, the error does not says much 
really, just that either the
other end timed out expecting a response from you, or your end timed
out expecting a response from the other end :)

However, from my experience, it may be an error in your DNIS/ANI
configuration and/or an mfcr2 library error ( less likely but still
possible ). Anyway, you can lear how to debug this problems with this
little document I wrote:

http://www.moythreads.com/unicall/mfcr2-asterisk-unicall-0.2-english.pdf

Good Look and happy debugging! :)

- Moisés Silva


 
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Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

[BJ Weschke]

Correct. Because Ethernet cross-over cables are crossing over 1,2,3
and 6; no 1,2,4 and  5.

On 3/18/07, Jeronimo Romero <[EMAIL PROTECTED]> wrote:

So a regular cross over cable wouldn't work?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Sunday, March 18, 2007 10:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

 Yes. At that point, you're looking for a T1 "cross-over".

 The pinout is as follows:

1

4

RX/Ring/- <-->TX/Ring/-
2

5

RX/Tip/+ <-->TX/Tip/+
4

1

TX/Ring/- <-->RX/Ring/-
5

2

TX/Tip/+ <-->RX/Tip/+
3

3

Shield/Return/Ground
6

6

Shield/Return/Ground

On 3/18/07, Jeronimo Romero <[EMAIL PROTECTED]> wrote:
> I assume that I would need to cross these pins over if I were going
from
> t1 card to t1 card. Is this correct?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tom
> Sent: Sunday, March 18, 2007 7:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards
>
> A common Cat5 straight through cable will work fine.
>
> T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for
> signals.
>
> A T1 loopback plug would be wired 1 to 4 and 2 to 5.
>
> They come in handy for testing T1 cards or for providing a hard loop
> for the telco.
>
> Tom
>
> At 05:42 PM 3/18/2007, you wrote:
> >Is there any technical difference between a T1 cable and a cat5e
patch
> >cable as far as using them with Digium T1/E1 cards? Can PRI circuits
> >terminating at a smart jack connect successfully to Digium cards
using
> >straight through CAT5e cables? If so, are they using all of the pins
in
> >the cable?
> >
> >Thanks in advance
> >___
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> >To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [asterisk-users] camp on off-line phone

[Jeronimo Romero]

It would be cool if you could add some kind of login script capability to nodes 
in  sip.conf and iax.conf.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: Sunday, March 18, 2007 11:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] camp on off-line phone

Leif Neland wrote:

> When phone A registers, I want phone B to ring, when picked up, it should 
> call phone A and connect the phones.
> 
> Translated: When GF in Mexico powers up laptop where soft iax-phone 
> registers automatically, I want to talk to her asap :-)
> 
> How to?

I don't really know how to do this, but wouldn't it be easiest
if she just called you as soon as she is online?

Sorry for not being of any help. :-(


Regards,
  Philipp

-- 
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 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] camp on off-line phone

[Philipp Kempgen]

Leif Neland wrote:

> When phone A registers, I want phone B to ring, when picked up, it should 
> call phone A and connect the phones.
> 
> Translated: When GF in Mexico powers up laptop where soft iax-phone 
> registers automatically, I want to talk to her asap :-)
> 
> How to?

I don't really know how to do this, but wouldn't it be easiest
if she just called you as soon as she is online?

Sorry for not being of any help. :-(


Regards,
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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RE: [asterisk-users] T1 cable for Digium T1/E1 Cards

[Jeronimo Romero]

So a regular cross over cable wouldn't work?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Sunday, March 18, 2007 10:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

 Yes. At that point, you're looking for a T1 "cross-over".

 The pinout is as follows:

1

4

RX/Ring/- <-->TX/Ring/-
2

5

RX/Tip/+ <-->TX/Tip/+
4

1

TX/Ring/- <-->RX/Ring/-
5

2

TX/Tip/+ <-->RX/Tip/+
3

3

Shield/Return/Ground
6

6

Shield/Return/Ground

On 3/18/07, Jeronimo Romero <[EMAIL PROTECTED]> wrote:
> I assume that I would need to cross these pins over if I were going
from
> t1 card to t1 card. Is this correct?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tom
> Sent: Sunday, March 18, 2007 7:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards
>
> A common Cat5 straight through cable will work fine.
>
> T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for
> signals.
>
> A T1 loopback plug would be wired 1 to 4 and 2 to 5.
>
> They come in handy for testing T1 cards or for providing a hard loop
> for the telco.
>
> Tom
>
> At 05:42 PM 3/18/2007, you wrote:
> >Is there any technical difference between a T1 cable and a cat5e
patch
> >cable as far as using them with Digium T1/E1 cards? Can PRI circuits
> >terminating at a smart jack connect successfully to Digium cards
using
> >straight through CAT5e cables? If so, are they using all of the pins
in
> >the cable?
> >
> >Thanks in advance
> >___
> >--Bandwidth and Colocation provided by Easynews.com --
> >
> >asterisk-users mailing list
> >To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

[BJ Weschke]

Yes. At that point, you're looking for a T1 "cross-over".

The pinout is as follows:

1

4

RX/Ring/- <-->TX/Ring/-
2

5

RX/Tip/+ <-->TX/Tip/+
4

1

TX/Ring/- <-->RX/Ring/-
5

2

TX/Tip/+ <-->RX/Tip/+
3

3

Shield/Return/Ground
6

6

Shield/Return/Ground

On 3/18/07, Jeronimo Romero <[EMAIL PROTECTED]> wrote:

I assume that I would need to cross these pins over if I were going from
t1 card to t1 card. Is this correct?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Sent: Sunday, March 18, 2007 7:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

A common Cat5 straight through cable will work fine.

T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for
signals.

A T1 loopback plug would be wired 1 to 4 and 2 to 5.

They come in handy for testing T1 cards or for providing a hard loop
for the telco.

Tom

At 05:42 PM 3/18/2007, you wrote:
>Is there any technical difference between a T1 cable and a cat5e patch
>cable as far as using them with Digium T1/E1 cards? Can PRI circuits
>terminating at a smart jack connect successfully to Digium cards using
>straight through CAT5e cables? If so, are they using all of the pins in
>the cable?
>
>Thanks in advance
>___
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>To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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RE: [asterisk-users] T1 cable for Digium T1/E1 Cards

[Jeronimo Romero]

I assume that I would need to cross these pins over if I were going from
t1 card to t1 card. Is this correct?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Sent: Sunday, March 18, 2007 7:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

A common Cat5 straight through cable will work fine.

T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for
signals.

A T1 loopback plug would be wired 1 to 4 and 2 to 5.

They come in handy for testing T1 cards or for providing a hard loop 
for the telco.

Tom

At 05:42 PM 3/18/2007, you wrote:
>Is there any technical difference between a T1 cable and a cat5e patch
>cable as far as using them with Digium T1/E1 cards? Can PRI circuits
>terminating at a smart jack connect successfully to Digium cards using
>straight through CAT5e cables? If so, are they using all of the pins in
>the cable?
>
>Thanks in advance
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[asterisk-users] camp on off-line phone

[Leif Neland]

When phone A registers, I want phone B to ring, when picked up, it should 
call phone A and connect the phones.


Translated: When GF in Mexico powers up laptop where soft iax-phone 
registers automatically, I want to talk to her asap :-)


How to?

Leif

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Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

[Tom]

A common Cat5 straight through cable will work fine.

T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for signals.

A T1 loopback plug would be wired 1 to 4 and 2 to 5.

They come in handy for testing T1 cards or for providing a hard loop 
for the telco.


Tom

At 05:42 PM 3/18/2007, you wrote:

Is there any technical difference between a T1 cable and a cat5e patch
cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards using
straight through CAT5e cables? If so, are they using all of the pins in
the cable?

Thanks in advance
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Re: [asterisk-users] T1 cable for Digium T1/E1 Cards

[BJ Weschke]

You can use a patch cable, yes. The T1 will look to use pins 1,2,4
and 5 while Ethernet will typically use 1,2,3 and 6 provided you're
not using POE or something simliar that requires additional pins.

On 3/18/07, Jeronimo Romero <[EMAIL PROTECTED]> wrote:

Is there any technical difference between a T1 cable and a cat5e patch
cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards using
straight through CAT5e cables? If so, are they using all of the pins in
the cable?

Thanks in advance
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--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] T1 cable for Digium T1/E1 Cards

[Jeronimo Romero]

Is there any technical difference between a T1 cable and a cat5e patch
cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards using
straight through CAT5e cables? If so, are they using all of the pins in
the cable?

Thanks in advance
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Re: [asterisk-users] Re: Replacement Wiki - options (Formerly 'status of voip-info')

[Davis Sylvester III]

Michiel ten Hagen wrote:
I also see the need for an asterisk only wiki. 


That is the reason I have started one at http://www.asterisk-wiki.org/
the information from voip-info.org is being added now as a starting
point. 


Ofcourse the success of this wiki will depends on the information
posted.

Regards,

Michiel ten Hagen

On Thu, 2007-03-15 at 16:05 -0600, Ira Burton wrote:
  

I would take an alternative stance and say that an Asterisk only
solution is needed.
This is a wildly growing product with nearly limitless
possibilities.  
Trying to cram too much on a site just causes confusion. 


KISS (no I am not calling anybody in particular stupid.)

On 3/15/07, Davis Sylvester III <[EMAIL PROTECTED]> wrote:
Michael Collins wrote:
>> I would suggest that we create a new wiki, make it solely
for Asterisk 
>> topics, as not to offend or replace voip-info.  Build

mirrors to
>> multiple sites and multiple domain names.  This would give
this
>> community a second resource with redundancy which is what I
think ALL 
>>

> of
>
>> us are looking for.  I have taken the pleasure, of
registering the
>> domain name ASTERISKONLINE.ORG.
>>
>
> I would like to know what the community feels about an
Asterisk-only 
> wiki.  I can see pros and cons of Asterisk-only vs.

> Asterisk/FreeSwitch/Yate/OpenPBX/etc.  My gut says keep it
open for
> everything OSS/VoIP.  (I have no logical reason for feeling
that way -
> it's just a gut feeling.) 
>

>
>
>> I will donate a dedicated server with bandwidth to the
cause.  I am
>> looking for additional people to help populate the wiki
with useful
>> information and to help maintain the site.  I would suggest
that ee 
>>

> have
>
>> maybe 4 or 5 mirrors to start off and a core group of
admins to help
>> maintain the site.
>>
>
> Thanks for putting your money where your mouth is!  This is
the kind of 
> action the community needs.

>
>
>> I am willing to work with anyone else that is about
providing a
>>
> solution
>
>> to our current issue.  If you guys want to REALLY work
toward a 
>> solution, here's the chance.  For the individuals that are

interested
>>
> in
>
>> helping e-mail me.
>>
>
> I hope you get some respondents.  In the meantime it might
be good to 
> check out the fledgling wiki here:

> http://www.voip-wiki.us
>
> It uses MediaWiki which has a nice, clean interface and
seems pretty
> easy to use.
>
> -MC
>
>
>
>
I'm okay with OSS/VoIP.  Just need confirmation that we all
want to do
this.  I don't want to allocate a server to the cause and it
just sit
idle.  I'm willing to work with the guys with
www.voip-wiki.us.

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Guys, I understand the reasoning, but don't you think that we have had 
enough volunteers for a asterisk-Only wiki / site for information.  It 
is my opinion that if we keep registering domain name and keep trying to 
keep multiple areas for resources, the community will have to idea where 
to look for accurate and up-to-date information.


I will be the 1st to offer my domain name to support a single site/cause 
for asterisk only information.  That doesn't mean we can't have multiple 
mirrors, but the information and site must be the same.


I would suggest that we join forces and start putting this together as a 
team, instead of as individuals.  If anyone wants to join forces to form 
this team, let me know and let's get it moving forward.


It this point I don't care what the domain name is, lets just work 
together for a common cause.


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Re: [asterisk-users] SMS Integration and SMS commands

[younss azzayani]

may be this will help :)
http://www.celliax.org
Good luck
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Re: [asterisk-users] Call counter for sip misbehaving

[Olle E Johansson]

17 mar 2007 kl. 12.21 skrev Rizwan Hisham:


Hi,
I have declared my sip users call-limit=2 and type=friend. When any  
user recieves a waiting call while already in a conversation, the  
peer call counter is set to 2.The problem is that, the counter is  
not reset to zero after hangup and becoz of this the user is not  
able to recieve any call anymore even if s/he has hungup. the  
asterisk cli displays the following error.


[Mar 17 16:15:10] ERROR[7664]: chan_sip.c:3030 update_call_counter:  
Call to peer 'rehmat' rejected due to usage limit of 2

-- Couldn't call rehmat
  == Everyone is busy/congested at this time (0:0/0/0)

Im using asterisk1.4.0 . declaring type=peer solves the problem.  
but if anybody knows why its not working for type=friend, plz share.



Have you read sip.conf.sample?

;call-limit=1			; permit only 1 outgoing call and 1 incoming call at  
a time

; from the phone to asterisk
; 1 for the explicit peer, 1 for the explicit 
user,
; remember that a friend equals 1 peer and 1 
user in
; memory
; This will affect your subscriptions as well.
; There is no combined call counter for a 
"friend"
; so there's currently no way in sip.conf to 
limit
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.

We do document things now and then and expect users to read the  
documentation :-)


Also please check
http://lists.digium.com/pipermail/asterisk-dev/2007-February/026190.html

/Olle


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[asterisk-users] Re: Call counter for sip misbehaving

[Tomislav Parcina]

Rizwan Hisham wrote:
Im using asterisk1.4.0 . declaring type=peer solves the problem. but if 
anybody knows why its not working for type=friend, plz share.


Please try 1.4.1, this should be fixed.


--
Tomislav Parcina
[EMAIL PROTECTED]

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[asterisk-users] Choppy sound with chan_capi + Fritz Card USB

[asterisk-users]

Hi everybody,

I have a problem which I cannot eliminate on my own. Has anybody any idea
for the following:

I am using the asterisk-version from Debian-Testing (1.2.13) with the
latest chan_capi (also tried an older version). 

When using the Capi-Channel, everything works fine except from the sound
it sounds extremely choppy and is unusable :-(

When e.g. capisuite is used for fax, everything sounds fine...

I found the following when using capi debug:

ISDN1#02: too much voice to send for NCCI=0x10101

Google finds nothing relevant for this error message :-(

Has anybody any idea ?

Christoph

P.S.: Here is the output of capi debug

CONNECT_IND ID=002 #0x016e LEN=0037
  Controller/PLCI/NCCI= 0x101
  CIPValue= 0x10
  CalledPartyNumber   = XXX
  CallingPartyNumber  = <00 a3>
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = <80 90 a3>
  LLC = default
  HLC = <91 81>
  AdditionalInfo  = default

-- CONNECT_IND (PLCI=0x101,DID=XXX,CID=,CIP=0x10,CONTROLLER=0x1)
   > ISDN1#02: msn='*' DNID='XXX' MSN
  == ISDN1#02: setting format alaw - 0x8 (alaw)
  == ISDN1#02: Incoming call '' -> 'XXX'
INFO_IND ID=002 #0x016f LEN=0017
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x1e
  InfoElement = <80 83>

INFO_RESP ID=002 #0x016f LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1#02: info element PI 80 83
   > ISDN1#02: Origination is non ISDN
INFO_IND ID=002 #0x0170 LEN=0022
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x70
  InfoElement = XXX

INFO_RESP ID=002 #0x0170 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1#02: info element CALLED PARTY NUMBER
   > ISDN1#02: INFO_IND DID digits not used in this state.
INFO_IND ID=002 #0x0171 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = <8a>

INFO_RESP ID=002 #0x0171 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1#02: info element CHANNEL IDENTIFICATION 8a
INFO_IND ID=002 #0x0172 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0xa1
  InfoElement = 

INFO_RESP ID=002 #0x0172 LEN=0012
  Controller/PLCI/NCCI= 0x101

-- ISDN1#02: info element Sending Complete
-- ISDN1#02: CAPI/ISDN1/XXX-3: XXX matches in context external
-- Executing VoiceMail("CAPI/ISDN1/XXX-3", "1234") in new stack
  == ISDN1#02: Answering for XXX
CONNECT_RESP ID=002 #0x016e LEN=0042
  Controller/PLCI/NCCI= 0x101
  Reject  = 0x0
  BProtocol
   B1protocol = 0x1
   B2protocol = 0x1
   B3protocol = 0x0
   B1configuration= default
   B2configuration= default
   B3configuration= default
   GlobalConfiguration= default
  ConnectedNumber = <00 80>XXX
  ConnectedSubaddress = default
  LLC = default
  AdditionalInfo
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default
   SendingComplete= default

-- Playing 'vm-intro' (language 'de')
  == Started pbx on channel CAPI/ISDN1/XXX-3
   > CAPI devicestate requested for ISDN1/XXX
   > CAPI devicestate requested for ISDN1/XXX
CONNECT_ACTIVE_IND ID=002 #0x0175 LEN=0015
  Controller/PLCI/NCCI= 0x101
  ConnectedNumber = default
  ConnectedSubaddress = default
  LLC = default

CONNECT_ACTIVE_RESP ID=002 #0x0175 LEN=0012
  Controller/PLCI/NCCI= 0x101

CONNECT_B3_IND ID=002 #0x0176 LEN=0013
  Controller/PLCI/NCCI= 0x10101
  NCPI= default

CONNECT_B3_RESP ID=002 #0x0176 LEN=0015
  Controller/PLCI/NCCI= 0x10101
  Reject  = 0x0
  NCPI= default

CONNECT_B3_ACTIVE_IND ID=002 #0x0177 LEN=0013
  Controller/PLCI/NCCI= 0x10101
  NCPI= default

CONNECT_B3_ACTIVE_RESP ID=002 #0x0177 LEN=0012
  Controller/PLCI/NCCI= 0x10101

DATA_B3_CONF ID=002 #0x0143 LEN=0016
  Controller/PLCI/NCCI= 0x10101
  DataHandle  = 0x13a
  Info= 0x0

DATA_B3_REQ ID=002 #0x0143 LEN=0030
  Controller/PLCI/NCCI= 0x10101
  Data32  = 0x8168df4
  DataLength  = 0xa0
  DataHandle  = 0x13a
  Flags   = 0x0
  Data64  = 0x0

DATA_B3_RE

[asterisk-users] Conference server (or how to make a call with more than 3 users)

[Yehavi Bourvine +972-8-9489444]

Hello,


  On most SIP phones a conference call is done on the phone and is limited to 3
participants. Polycom phones has a configuration option to use a conference
server instead of the internal conferencing feature. I guess I need some
conference server; any experience with such a server which can interact with
Asterisk?

Thanks, __Yehavi:
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RE: [asterisk-users] Follow me on multiple numbers..

[Kevin Kiely]

I tried to look at the code in Trixbox but when the option 'confirm' is
selected in the follow me properties screen, no code is generated and the
call goes dead.  Is there a trick to get the code generated?
 
 
  _  

From: Philippe Lindheimer [mailto:[EMAIL PROTECTED] 
Sent: Saturday, March 17, 2007 12:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Follow me on multiple numbers..
 
On 3/16/07, Ritesh Agrawal <[EMAIL PROTECTED]> wrote:
> Hi Folks,


>


> I want to setup a follow me routine so that asterisk can call me on the


> multiple numbers.


> I tried some of the samples at voip-info but there is a problem with those


> examples.


>


> I dont have coverage in my home area and my cell phone answering machine


> picks up the phone right away so my home phone never rings.


> I also want the caller to be able to leave a voicemail and the cell phone


> answering machine messes it all up.


> I have call screening setup so the call gets answered by the cell phone


> answering machine and it never accepts the call.


>


> I would appreciate if someone can help me with the setup.


>


You can create a follow-me
 with 1.2 that requires you to confirm the call before


answering the channel. If you need an example, go have a look at the code I


generate in the dialplan in freepbx to do that exact thing when you choose
call


confirm. No need to go to 1.4 just for that.





philippel
  
  _  

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  miss an email again!
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[asterisk-users] Re: Only secretary can call the boss, all others only reach the secretary when dial the boss extension

[Benny Amorsen]

> "RC" == Ricardo Carvalho <[EMAIL PROTECTED]> writes:

RC> I've also tried to do it using different contexts, but it still
RC> doesn't work. I've done like this:

RC> [default]
RC> exten => secretary_extension,1,Dial(SIP/secretary_extension)
RC> exten => boss_extension,1,Dial(SIP/secretary_extension)
RC> [secretary]
RC> include => default
RC> exten => boss_extension,1,Dial(SIP/boss_extension)

That really should have worked. Can the secretary call the boss at all
with that setup? If so, attended transfers will definitely work --
asterisk can't tell the difference between an attended transfer and a
plain call until after the call is placed. Unattended transfers should
work too, and even putting a redirect on the secretary phone should
work.


/Benny


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