date:20060126


Damon Estep a écrit :

Can anyone point me to a reference or sample config for bypassing a 
nailed up (point to point) t1 between two PBXs with asterisk and a 
pair of t1 cards?


 

Right now I have 2 Nortel norstars connected to each other via a 
leased line t1. I also have a solid 10mbps low latency microwave link 
between the 2 sites.


You probably need a couple of T1 cards, and some paid consulting to get 
it working (I've never done it myself but that's how I would do it if I 
was in a hurry)



My goal is to run an asterisk box at each end with a t1 card and 
Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in 
a relatively dumb configuration, with the goal of migrating off of the 
norstars eventually.


If it's a point to point Asterisk - Asterisk configuration, why use 
SIP rather than IAX? IAX configuration is very easy, so once you get the 
norstar - asterisk link up it'll be a piece of cake.


Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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RE: [Asterisk-Users] * point to point t1 solution?

Jean-Michel,

You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING 
A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I 
MIGHT LOOK AT?  Not WILL YOU DO IT FOR ME?

Your response to this post was un-informative and quite frankly it is the type 
of useless response that most mailing lists and newsgroups could do without.

Damon

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
 Sent: Thursday, January 26, 2006 1:36 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution?
 
 Damon Estep a écrit :
 
  Can anyone point me to a reference or sample config for bypassing a
  nailed up (point to point) t1 between two PBXs with asterisk and a
  pair of t1 cards?
 
 
 
  Right now I have 2 Nortel norstars connected to each other via a
  leased line t1. I also have a solid 10mbps low latency microwave link
  between the 2 sites.
 
 You probably need a couple of T1 cards, and some paid consulting to get
 it working (I've never done it myself but that's how I would do it if I
 was in a hurry)
 
 
  My goal is to run an asterisk box at each end with a t1 card and
  Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in
  a relatively dumb configuration, with the goal of migrating off of the
  norstars eventually.
 
 If it's a point to point Asterisk - Asterisk configuration, why use
 SIP rather than IAX? IAX configuration is very easy, so once you get the
 norstar - asterisk link up it'll be a piece of cake.
 
 Cheers,
 Jean-Michel.
 
 --
 Jean-Michel Hiver - http://ykoz.net/
 Découvrez la Réunion des Technologies IP  Telecom
 TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
 
 
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Re: [Asterisk-Users] * point to point t1 solution?

2006-01-26 Thread Simon Woodhead

Bad day Damon? I think your comments are a little harsh towards someone who is an active and informed contributor to the list. Jean-Michel could have ignored you but he chose to share what he could. Maybe someone else will have the complete answer to your question.
On 1/26/06, Damon Estep [EMAIL PROTECTED] wrote:
Jean-Michel,You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I MIGHT LOOK AT?Not WILL YOU DO IT FOR ME?
Your response to this post was un-informative and quite frankly it is the type of useless response that most mailing lists and newsgroups could do without.Damon -Original Message- From: 
[EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]
] On Behalf Of Jean-Michel Hiver Sent: Thursday, January 26, 2006 1:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * point to point t1 solution?
 Damon Estep a écrit :  Can anyone point me to a reference or sample config for bypassing a  nailed up (point to point) t1 between two PBXs with asterisk and a  pair of t1 cards?
 Right now I have 2 Nortel norstars connected to each other via a  leased line t1. I also have a solid 10mbps low latency microwave link  between the 2 sites.
  You probably need a couple of T1 cards, and some paid consulting to get it working (I've never done it myself but that's how I would do it if I was in a hurry)  My goal is to run an asterisk box at each end with a t1 card and
  Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in  a relatively dumb configuration, with the goal of migrating off of the  norstars eventually. 
 If it's a point to point Asterisk - Asterisk configuration, why use SIP rather than IAX? IAX configuration is very easy, so once you get the norstar - asterisk link up it'll be a piece of cake.
 Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP  Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
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Re: [Asterisk-Users] * point to point t1 solution?

2006-01-26 Thread Adrian Carter





Damon,
 I am not intimately familiar with what you are specifically trying
to achieve, *BUT*, if the two Norstars are essentially just
'interconnected' via teh T1 to provide either an EM Wink "type"
connection/private TDM bus between the two boxes so that extensions are
'bridged' between the two PABX's.. then *YES*.. Asterisk can do that.

 I have a setup at present of an Asterisk box that has 1 E1 PRI
coming in from a telco, and 1 E1 PRI going TO an Ericsson BP250 system.
The Asterisk box transparently passes incoming calls destined for the
BP250 to the BP250, and all the BP250 users can 'direct-dial' so to
speak Asterisk extensions and vice-versa.

 Since your original question is a tad ambigious to someone not
entirely intimate with your setup, I could have missed the point
entirely, as there is quite a few 'ways' to tie PABX's together and the
reasoning can be very different (Could be for 'switched' extensions,
could be for LCR setup, could be for VoiceMail integration, could be
cause thats what the original installer just did...)

 Hope that helps you a little. I'd suggest looking at "Connecting
Asterisk to Legacy PABX's" and then extrapolate that information to
suit your needs.

Regards

Adrian

Damon Estep wrote:

  Jean-Michel,

You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I MIGHT LOOK AT?  Not WILL YOU DO IT FOR ME?

Your response to this post was un-informative and quite frankly it is the type of useless response that most mailing lists and newsgroups could do without.

Damon

  
  
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]] On Behalf Of Jean-Michel Hiver
Sent: Thursday, January 26, 2006 1:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * point to point t1 solution?

Damon Estep a crit :



  Can anyone point me to a reference or sample config for bypassing a
nailed up (point to point) t1 between two PBXs with asterisk and a
pair of t1 cards?



Right now I have 2 Nortel norstars connected to each other via a
leased line t1. I also have a solid 10mbps low latency microwave link
between the 2 sites.

  

You probably need a couple of T1 cards, and some paid consulting to get
it working (I've never done it myself but that's how I would do it if I
was in a hurry)




  My goal is to run an asterisk box at each end with a t1 card and
Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in
a relatively dumb configuration, with the goal of migrating off of the
norstars eventually.

  

If it's a point to point Asterisk - Asterisk configuration, why use
SIP rather than IAX? IAX configuration is very easy, so once you get the
norstar - asterisk link up it'll be a piece of cake.

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Dcouvrez la Runion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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-- 
Adrian Carter
Technical Manager
Leading Edge Internet

Web	  http://www.lei.net.au http://support.lei.net.au
Direct+61 2 6163 6162  Support 1 300 662 415
E-mail[EMAIL PROTECTED]


-- 
Adrian Carter
Technical Manager
Leading Edge Internet

Web	  http://www.lei.net.au http://support.lei.net.au
Direct+61 2 6163 6162  Support 1 300 662 415
E-mail[EMAIL PROTECTED]



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[Asterisk-Users] TDM400 pinout

2006-01-26 Thread bails

Hi I'm looking for a pinout for the above.  Note this has what i'd call 
RJ45 sockets (or someone smart can correct me).  I need to plug into BT 
(rj13?).


And, yes I've googled (glad I'm not chinese) and have tried the 
suggested, just plug in a 6 connector rj11 and i didnt work atall.


On a side note, I cannot beleive that Digium dont have this information 
on there site, it seems somewhat lacking as the only info i can find on 
there site about this board is a pretty PDF.


Also, I 'm still having problems with another one of these boards not 
dropping the line ([EMAIL PROTECTED] 1.5, plesae dont say post the [EMAIL PROTECTED] forum). 
Any Ideas?


thanks in advance

Bails
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Re: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service

2006-01-26 Thread Chris Stenton

I have been using sipdiscount in sip mode (they are discontinuing their IAX2 
connection) for a while. UK calls are free and its worked most of the time. 
However, its not working this morning .-(

Chris

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

Sent: Thursday, January 26, 2006 6:35 AM
Subject: Re: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service

I tried through voipdiscount as well.
Even my older account through voipbuster started to behave this way and it 
used to be ok on IAX.

I would expect at least some reply.

Rudolf

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

Sent: Thursday, January 26, 2006 5:00 PM
Subject: Re: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service

Didn't you read this from their QA?

I want to configure my own IAX/SIP device for calling with VoipBuster, is 
that possible?
It is possible to use your own IAX/SIP device, however we do not support 
it. We advise you to use SIP-Discount instead.

Do you have the same problem when you use their softphone? If not, why 
complaining.

The call to the UK is free only for VoIPstunt

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

Sent: Thursday, January 26, 2006 6:19 AM
Subject: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service

Hi, all

I am reallty pissed with their service. I wonder if this is common 
problem.
Firstly, all of my calls are terminated after 30s. And termination 
happens
in a strange way. My local asterisk server does not see the 
disconnection,
but remote party is disconnected. Basically, I am still on the phone, 
while

remote party was disconnected. When I hang up, I get something like that:

Apr 20 02:32:43 WARNING[4853]: chan_sip.c:8520 handle_response: Got
authentication request (401) on unknown BYE to
'sip:[EMAIL PROTECTED];tag=c9ebef50c90078c2c93eddc243d7352d6e04'

Secondly, they charged me for calls to UK that was supposed to be free.
And their customer service does not respond at all. Do they have a phone
number I can call?

Thanks,
Rudolf

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RE: [Asterisk-Users] * point to point t1 solution?









Actually, it is a quite appropriate
response to ANYONE that includes this type of comment in their reply



You probably need a
couple of T1 cards, and some paid consulting to get it working (I've never done
it myself but that's how I would do it if I was in a hurry)



Perhaps something like this would have been better received;



I know it can (or cannot) be done, and here is the name of
someone that might be willing to help you for a fee



Look back though the archives and you will see that I have had some participation
here myself  in the past



D













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Woodhead
Sent: Thursday, January 26, 2006
2:01 AM
To: Asterisk Users Mailing List -
 Non-Commercial Discussion
Subject: Re: [Asterisk-Users] *
point to point t1 solution?





Bad day Damon? I think
your comments are a little harsh towards someone who is an active and informed
contributor to the list. Jean-Michel could have ignored you but he chose to
share what he could. Maybe someone else will have the complete answer to your
question. 



On 1/26/06, Damon
 Estep [EMAIL PROTECTED]
wrote:

Jean-Michel,

You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING
A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I
MIGHT LOOK AT?Not WILL YOU DO IT FOR ME?

Your response to this post was un-informative and quite frankly it is the type
of useless response that most mailing lists and newsgroups could do without.

Damon

 -Original Message-
 From: [EMAIL PROTECTED]
[mailto:asterisk-users-
 [EMAIL PROTECTED] ]
On Behalf Of Jean-Michel Hiver
 Sent: Thursday, January 26, 2006 1:36 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution? 

 Damon Estep a écrit :

  Can anyone point me to a reference or sample config for bypassing a
  nailed up (point to point) t1 between two PBXs with asterisk and a
  pair of t1 cards? 
 
 
 
  Right now I have 2 Nortel norstars connected to each other via a
  leased line t1. I also have a solid 10mbps low latency microwave link
  between the 2 sites. 
 
 You probably need a couple of T1 cards, and some paid consulting to get
 it working (I've never done it myself but that's how I would do it if I
 was in a hurry)


  My goal is to run an asterisk box at each end with a t1 card and 
  Ethernet card to act as a TDMSIP gateway to bypass the nailed
T1 in
  a relatively dumb configuration, with the goal of migrating off of
the
  norstars eventually.
 
 If it's a point to point Asterisk - Asterisk configuration, why
use
 SIP rather than IAX? IAX configuration is very easy, so once you get the
 norstar - asterisk link up it'll be a piece of cake. 

 Cheers,
 Jean-Michel.

 --
 Jean-Michel Hiver - http://ykoz.net/
 Découvrez la Réunion des Technologies IP  Telecom
 TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE 


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[Asterisk-Users] codec selection based on call prefix

2006-01-26 Thread Dionisis Koumouras




Hi all,
Ihave an IAX connection between two asterisk 
servers and i'm looking for a way to cut down on the needed bandwidth. Both 
voice and fax calls pass through the channel so it is currently configured to 
use g.711.Could it be possible to select the codec based on the call's 
prefix so that g.711 will be used for fax calls and g.729 for 
voice?

Dionisis
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Re: [Asterisk-Users] * point to point t1 solution?


Damon Estep a écrit :


Jean-Michel,

You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING 
A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I 
MIGHT LOOK AT?  Not WILL YOU DO IT FOR ME?
 

Yes, I think Asterisk can do what you are trying to achieve. No, I don't 
know how to do it, and no I won't do it for you since I've never been in 
the situation you're in.


As for web based hints, with some experience I've found that google is 
as good as asking the mailing list. If you have no success with the 
mailing list, the wiki (voip-info.org) and google, you can try the irc 
channel #asterisk on freenode.


If that still doesn't work, shelling out a few hundred bucks for a 
consultant to help you do it - and train you in the process - is the 
other alternative, and is often a good deal.


I've done it a couple of times myself, and it's awesome how far people 
can get you and how hard they try when you recognize the value of their 
work with some money (as opposed to just asking nicely).


Cheers,
Jean-Michel.

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Re: [Asterisk-Users] VOIP Router

2006-01-26 Thread Kristian Larsson

On Thu, Jan 26, 2006 at 09:42:36AM +0200, Mohamed Farid wrote:
 Dear All :
 I need to link my HQ to some Remote Sites - I need a Router which
 supports VOIP , and VPN
 Also the Router Should has 3 FXS ports and 1 FXO ...
 The call should be routed from the Remote Site to the HQ through a VPN
 tunnel ( 3DES ) ...
 Any Advise ?
The cisco x8xx series are excellent. I have a
2811, if you're routing needs are basic a 2801
should suit you just find you can cram a few
VIC2-2FXS in it and get the voice ports you need.
It's capable of 3DES and comes in a nice package
too. An excellent router.
Oh, and it's rather cheap too :)

  Kristian.

-- 
Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
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Re: [Asterisk-Users] ACD with polycom ip phones

2006-01-26 Thread hgaillac-sip

Hello,

Can you provide a patch from your special branch for
asterisk-1.2.3 ?
can you post a how-to ?

Even these features won't be include in th main
branche a patch should be available.

Regards
harry

--- BJ Weschke [EMAIL PROTECTED] a écrit :

  On 1/25/06, Douglas Garstang
 [EMAIL PROTECTED] wrote:
  I've tried that. Setting acd-login-logout and
 acd-agent-available to 1 causes the appearance to
 automatically log in when the phones comes up, and
 stays up the entire time. I'll have another shot it
 in a bit tho maybe I missed something before.
 
 
  You need the code on that special branch in
 conjunction with the
 config setting in order for it to work.
 --
 Bird's The Word Technologies, Inc.
 http://www.btwtech.com/
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Re: [Asterisk-Users] Bootable CD?

2006-01-26 Thread Tzafrir Cohen

On Thu, Jan 26, 2006 at 01:02:09PM +0500, Sohail Arham wrote:
 hi ,
 i have downloaded the [EMAIL PROTECTED] software from the web ..but i have a
 little confusion about that ...either i wrote in blank cd as it is or some
 bootable media is required for it...as it is in zip format...BUT it is a
 .ISO file ...tell me ...what should i do...it will run automatically when i
 reboot system and first boot device is CDROm...thank

http://linuxiso.org/viewdoc.php/howtoburn.html

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend

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RE: [Asterisk-Users] * point to point t1 solution?

Jean-Michel,

I agree with all of your comments, and would be willing to bet $100 that NO 
AMOUNT OF GOOGLING will answer this question definitively.

After reviewing Adrian Carters very informative response regarding TDMoE I am 
getting closer to what I need to know (now my googles include asterisk AND 
TDMoE).

This is CLEARLY uncharted territory, while I'll bet it has been done before, no 
one took the time to document it.

My return to the list if I am successful will be to document the config on the 
wiki... fair exchange?

And, if along the way I find an EXPERT in this area with REAL WORLD PRODUCTION 
EXPERIENCE I will gladly pay the fees, but I am not about to shell out anything 
to pay some know-it-all to educate themselves and provide me a half baked 
solution that has never been put to the real world test.

D



 Damon Estep a écrit :
 
 Jean-Michel,
 
 You missed the entire point - the question is IS ASTERISK CAPABLE OF
 EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB
 BASED HINTS I MIGHT LOOK AT?  Not WILL YOU DO IT FOR ME?
 
 
 Yes, I think Asterisk can do what you are trying to achieve. No, I don't
 know how to do it, and no I won't do it for you since I've never been in
 the situation you're in.
 
 As for web based hints, with some experience I've found that google is
 as good as asking the mailing list. If you have no success with the
 mailing list, the wiki (voip-info.org) and google, you can try the irc
 channel #asterisk on freenode.
 
 If that still doesn't work, shelling out a few hundred bucks for a
 consultant to help you do it - and train you in the process - is the
 other alternative, and is often a good deal.
 
 I've done it a couple of times myself, and it's awesome how far people
 can get you and how hard they try when you recognize the value of their
 work with some money (as opposed to just asking nicely).
 
 Cheers,
 Jean-Michel.
 
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Re: [Asterisk-Users] Best FXO hardware for home use

2006-01-26 Thread Facundo Ameal

I'm using an X100P Clone at home and i had not much trouble, remember
I'm just testing and learning a bit at home. I think if you hace to
implement it at office you'll have to spend a bit more.

2006/1/25, Joseph Tanner [EMAIL PROTECTED]:
 Personally, I've had great success with an X101P (it's a clone, but
 it's the exact same chipset and layout of the original).  Now, with
 Asterisk 1.2 beta2 (I believe it was beta2, I could be wrong though)
 and a P3 933MHz PC I did get annoying echo that I couldn't get rid of,
 and only on outgoing calls.  If someone called me, even though all the
 same equipment is being used, there was no echo.  Anyways, I upgraded
 to [EMAIL PROTECTED] 2.2 with Asterisk 1.2.1 and at the same time upgraded
 to a Celeron 2.93GHz PC, and there's virtually no echo.  Only if
 there's complete silence on the other end and you yell very loud, can
 you barely make any hint of an echo out.  No idea if it was the
 Asterisk upgrade, the new PC, or both that fixed my problem.

 Also, somewhere around the pre-1.0 days, I had two of these clones
 (one was the exact same layout as the actual X101P, the other had a
 different layout but the same chipset) and the one I used with my
 Packet8 line had no echo, but my landline did.  Didn't matter if I
 switched the lines, the one connected to the Packet8 device had zero
 echo, the one connected to my landline had a noticeable echo (again,
 only on outgoing calls, incoming was fine).  Played with
 rxgain/txgain, all the echo settings, etc.  But now all is fine.

 Guess what I'm trying to say, is a lot depends on the line itself, and
 your exact setup.  If you can pick up an X101P clone for cheap, I'd
 try that first.  Most you're out is a few bucks (I say a few bucks,
 cause even if you pay $20 and decide it won't work for you, you can
 sell it for about what you paid).  If you build or repair PCs a lot
 for others, then you'll need a good cheap modem someday anyways, the
 clone cards work fine for that.

 Works fine for me, only issue I have now is callerid isn't 100%
 reliable, but works the majority of the time.  Until I troubleshoot it
 further (i.e., connect a regular phone directly to my landline to at
 least verify it's getting callerid when asterisk isn't), I can't blame
 the card for that.  As long as the card will work with your setup
 (odds are it will), I think it's the best solution for home or small
 business use.

 Joseph Tanner

 On 1/25/06, Rich Adamson [EMAIL PROTECTED] wrote:
 
echo cancellation is pretty limited on these cheap devices.
the spa3000 manual for example states the AEC is limited to
8ms. good AECs will handle 64ms or more. in my experience the
spa3000 echo canceller is cranky. it works most but not all
of the time.
  
   I have been using one for 6 months without any problems. Make sure you 
   have
   the most current firmware on it and it should work just fine.
 
  Kerry,
 
  There is an issue with the spa3k (as well as the TDM04b) in terms
  of handling echo properly on long pstn loops. You are obviously on
  a relatively short loop if you've not been exposed to the variable
  echo cancellation issues.
 
  In other words, long pstn loops basically fall outside the limits of
  the echo cancellation software as someone else already noted.
 
 
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Facundo Ameal.
famealatgmaildotcom
Linux User #395088

FWD: 741664
MSN: asadoatlamorcilladotcomdotar
ICQ: 74005793


Open your mind, use open source.
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RE: [Asterisk-Users] suggest a gsm router

2006-01-26 Thread Sam Tam









Why not try to purchase one of our GSM
Gateway at £60 and then you can route all the mobile calls through the GSM
Gateway?



http://cyber-telecom.net/store/product_info.php?products_id=29osCsid=4e787773c7c03212c43c51368d6ae387



Sam 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of amit chowrasia
Sent: Tuesday, January 24, 2006
5:18 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] suggest
a gsm router







Hi Everybody











I am building a small ippbx
network for my office





I have 6 hard ip phone's and
asterisk server but 





now for outging and incoming
calls i want to use





gsm router instead of x100p
card ... or pstn





I want my calls will
goout and comethrough mobile sim card (gsm router).





My mobile service provider
Simultaneously 64 lines conference...





How many calls can i receive
and outgo through my asterisk server at same time. with gsm router?





Can you suggest me a nice
gsm router with good price ?











Thank you





http://www.jrass.itgo.com














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RE: [Asterisk-Users] No audio? Update your Asterisk

Same situation.
Asterisk 1.2.1 ([EMAIL PROTECTED]  2.2) apparently doesn't have this problem.

Thanks
Mimmus


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Joseph Tanner
 Sent: Wednesday, January 25, 2006 4:05 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] No audio? Update your Asterisk
 
 For what it's worth, I've been messing around with my install 
 all night and haven't had a single issue.  [EMAIL PROTECTED] 2.2, 
 Asterisk version 1.2.1.  Even set the date ahead, still no 
 problems.  Could be a fluke, I'm interested if anyone else is 
 using 1.2.1 and has these issues, but for now I'm sticking 
 with what I have.

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Re: [Asterisk-Users] No audio? Update your Asterisk

2006-01-26 Thread Adam Goryachev

On Wed, 2006-01-25 at 14:10 -0600, Kevin P. Fleming wrote:
 Aaron Daniel wrote:
  We had the bug on 1.2.2, but when I rolled back to 1.2.1 to fix the 
  problem, everything started working.  Doesn't seem like it's a bug in 
  1.2.1 :)
 
 It is not. The bug was introduced during the 1.2.1-1.2.2 transition.

Observation ...

Had a problem with asterisk 1.2 trunk from approx 10 Dec, (from memory),
when the problem occurred, a restart of asterisk did not fix it, and a
reboot of the machine still did not fix it. A svn update and restart of
asterisk still did not fix it.

I wonder, was it really only 2^22 seconds, or how exactly did that
work?? I really was quite stumped when it happened, since nobody had
changed anything for ages...

Anyway, thankfully it broke at 4:30pm, and everyone went home at 5pm
(the after hours announce only and hangup was working), so I left it
until the next day. A few debug attempts, and a login to IRC, and
suddenly the solution was shouting at me (in the subject/title of the
IRC channel to svn update).

Mainly was confused as to why a asterisk + zaptel restart and a reboot
didn't fix the problem (even just temporarily)?

Regards,
Adam

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RE: [Asterisk-Users] Looking for Q.Sig success story

 Hi Mimmus, and thanks for the quick reply.
You are welcome.

 It is actually very good to hear that most of it works. The 
 difference in my project is that we'll keep the PSTN link on 
 the Alcatel, and use the asterisk only as a inter-site 
 trunking solution. The reason is that I have no Alcatel 
 knowledge (will rely on other people), and I want to be as 
 un-intrusive as possible. If you don't mind, I would have 
 some additional questions:
I have no knowledge of Alcatel too!
I think that putting Asterisk in front of Alcatel is the best way to offer *
advanced features (voicemail, audioconference, fax, ...) to all users.

 1) Can you confirm that Q.Sig is the only option for me ?
No idea!

 2) What hardware are you using on the Asterisk (Digium ?)
Tried both Digium TE410P and Sangoma A102. Better results with latest one.

 Once my pilot starts I may come back to you for some examples 
 and advice, but this probably won't happen before a month or so.
No problem.

Bye
Mimmus

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[Asterisk-Users] Good switchboard solution?

2006-01-26 Thread Roy Sigurd Karlsbakk


Hi

Does anyone know a good, scalable switchboard solution for asterisk?  
I've been looking around and I've found a couple but I'm not sure yet...


Have anyone here used one in large environments? We need usable GUI  
with the usual stuff like queues, transfer, meetme etc


roy
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Re: [Asterisk-Users] Digium hardware

2006-01-26 Thread pdhales




Move to PRI - it will be much more fun than working 
with analog.

PaulH


  - Original Message - 
  From: 
  Cisco - Kameko 
  
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, January 25, 2006 6:17 
  PM
  Subject: [Asterisk-Users] Digium 
  hardware
  
  Hello,
  
  I want to setup an asterisk pabx. I want to 
  understand more on what hardware (PCI cards) i will need to do this. I 
  have 5 xchange lines and 30 extensions within our offices. I have just 
  finished installing Fedora Core and downloaded asterisk-1.2.3.tar.gzand zaptel-1.2.2.tar.gzwhich i want to install.
  
  In need or your advise ASAP
  
  Regards,
  
  SOUL
  
  

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[Asterisk-Users] using sangoma cards as a timesource?

2006-01-26 Thread Roy Sigurd Karlsbakk


hi

building a new setup, we want to try using sangoma cards. can these  
be used as time sources the same way as TE410Ps?


thanks

roy
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[Asterisk-Users] Re: Random Disconnects

2006-01-26 Thread Tomislav Parcina

In article 77758c190601240743o3ae310dbi28b2f79a93965776
@mail.gmail.com, [EMAIL PROTECTED] says...
 I am not very satisfied with this, though.  I want to use some
 features (like Park) that apparently don't work well with reinvites. 
 Have any of the rest of you had any luck troubleshooting this problem?

Your RTP stream doesn't pass thrue Asterisk and it can't hear that you 
have pressed any key (that you are requesting that he parks the call).


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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RE: [Asterisk-Users] Asterisk + Ericsson PBX

Thanks!
You are welcome.

Now the E1 is up, but still problems.

What I'm trying to do, is to let calls arrive to Asterisk 
from the net, and using the Sangoma pass them to the PBX.
Is this possible?
Passing calls between different channels is the primary job of Asterisk, I
think! 
You have to write a dialplan but I cannot teach this here.

Bye
Mimmus

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RE: [Asterisk-Users] Digium hardware

2006-01-26 Thread Bogdan Moldovan

Hello,
 
The 5 exchange lines I assume they are analogic. For them you will need 5
FXO ports. You can buy a TDM04B and a TDM01B (this will get you to the 5
FXO). Make sure you have 2 PCI slots available.
 
Now for the extensions you need 
- IP Phones 
or
- ATAs (if you want to reuse your analog phones)
 
IP Phones you can buy from different companies. I like Aastra, Polycom...
 
For ATAs I like Linksys PAP2.
 
Bogdan Moldovan
VoIP SIP: sip://[EMAIL PROTECTED]
VoIP IAX: iax://obelisk.modulo.ro/101
MODULO Consulting
The Future Is Not What It Used To Be
http://www.modulo.ro 




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cisco - Kameko
Sent: Wednesday, January 25, 2006 9:17 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Digium hardware


Hello,
 
I want to setup an asterisk pabx. I want to understand more on what hardware
(PCI cards) i will need to do this. I have 5 xchange lines and 30
extensions within our offices. I have just finished installing Fedora Core
and downloaded asterisk-1.2.3.tar.gz
http://ftp.digium.com/pub/asterisk/asterisk-1.2.3.tar.gz  and
zaptel-1.2.2.tar.gz http://ftp.digium.com/pub/zaptel/zaptel-1.2.2.tar.gz
which i want to install.
 
In need or your advise ASAP
 
Regards,
 
SOUL

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[Asterisk-Users] Calls pickup

Hi,
is it possible pickup calls (with *8) between different channels (SIP and
IAX)?


Thanks
Mimmus

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Re: [Asterisk-Users] transfer, recording ...

2006-01-26 Thread Bartosz Piec


Ronald Wiplinger wrote:
I tried to transfer a call, pickupcall and onetouch recording, but have 
not got it to work!


You must uncomment the lines in feature.conf (remove the ; character 
from the beggining).


--
Best regards,
Bartosz Piec
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Re: [Asterisk-Users] ACD with polycom ip phones

2006-01-26 Thread BJ Weschke

On 1/26/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 Hello,

 Can you provide a patch from your special branch for
 asterisk-1.2.3 ?
 can you post a how-to ?

 Even these features won't be include in th main
 branche a patch should be available.


 Harry -

 There is a patch available against /trunk, not 1.2.3. As I said in an
earlier email, it will take a little more work to produce a patch that
compiles correctly against 1.2.3 as the code has changed a good bit.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[Asterisk-Users] 0h323 - one way audio

2006-01-26 Thread yusuf





I am using 0h323 on Asterisk CVS HEAD 19/07/2005.  I am dialling a h323 
gatekeeper.  He can hear me, but I cannot hear him.


I have a suspicion that it could be the rtp traffic, since he said that 
they need rtp traffic from ports 4500 - 65000.   So in 0h323.conf i set 
updstart and udpend, and in rtp.conf i set the ports here.


a tcpdump confirms there is two way traffic.
unfortunately, a 0h323 debug does not show much:

  -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-2, 
OH323/[EMAIL PROTECTED]) in new stack

-- H.323 call to [EMAIL PROTECTED]
-- Called [EMAIL PROTECTED]
-- OH323/[EMAIL PROTECTED] is ringing
ECNLONDON2*CLI oh323 debug toggle
Verbose debug info for OpenH323 channel driver turned on.
Channel OH323/[EMAIL PROTECTED] (call 'ip$localhost/13673') 
TX byte count is 4000.
Channel OH323/[EMAIL PROTECTED] (call 'ip$localhost/13673') 
RX byte count is 7000.
Channel OH323/[EMAIL PROTECTED] (call 'ip$localhost/13673') 
TX byte count is 5000.
Channel OH323/[EMAIL PROTECTED] (call 'ip$localhost/13673') 
RX byte count is 8000.

Channel OH323/[EMAIL PROTECTED] (call 'ip$localhost/13673')

do have any ideas?

thanks,
yusuf
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Re: Re: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service

2006-01-26 Thread aryanto.rachmad

Did you know that they switched over to a new set of servers? And they also 
planning to switch off IAX very soon (as per their email notification to me on 
the 13th of January)?

 
 Von: RumaTech [EMAIL PROTECTED]
 Datum: 2006/01/26 Do AM 07:35:49 CET
 An: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
 Betreff: Re: [Asterisk-Users] Voipbuster/voipstunt -- what a  crap service
 
 I tried through voipdiscount as well.
 Even my older account through voipbuster started to behave this way and it 
 used to be ok on IAX.
 
 I would expect at least some reply.
 
 Rudolf
 
 - Original Message - 
 From: Aryanto Rachmad [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, January 26, 2006 5:00 PM
 Subject: Re: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service
 
 
 Didn't you read this from their QA?
 
 I want to configure my own IAX/SIP device for calling with VoipBuster, is 
 that possible?
 It is possible to use your own IAX/SIP device, however we do not support it. 
 We advise you to use SIP-Discount instead.
 
 Do you have the same problem when you use their softphone? If not, why 
 complaining.
 
 The call to the UK is free only for VoIPstunt
 
 - Original Message - 
 From: RumaTech [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, January 26, 2006 6:19 AM
 Subject: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service
 
 
  Hi, all
 
  I am reallty pissed with their service. I wonder if this is common 
  problem.
  Firstly, all of my calls are terminated after 30s. And termination happens
  in a strange way. My local asterisk server does not see the disconnection,
  but remote party is disconnected. Basically, I am still on the phone, 
  while
  remote party was disconnected. When I hang up, I get something like that:
 
  Apr 20 02:32:43 WARNING[4853]: chan_sip.c:8520 handle_response: Got
  authentication request (401) on unknown BYE to
  'sip:[EMAIL PROTECTED];tag=c9ebef50c90078c2c93eddc243d7352d6e04'
 
  Secondly, they charged me for calls to UK that was supposed to be free.
  And their customer service does not respond at all. Do they have a phone
  number I can call?
 
  Thanks,
  Rudolf
 
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Re: [Asterisk-Users] Bootable CD?

2006-01-26 Thread Dovid Bender

When you open your burning software there should be an
option to burn from an image. When it asks you for the
location tof the image point it to the .iso file that
you downloaded. After it is done burning the CD you
have a ready to go bootable CD. BE CAREFULL. Once you
put the CD into a machine it will format the machine
and delete everything on it and install CentOS with
asterisk.

Regards,
Dovid

--- Sohail Arham [EMAIL PROTECTED] wrote:

 hi ,
 i have downloaded the [EMAIL PROTECTED] software from
 the web ..but i have a
 little confusion about that ...either i wrote in
 blank cd as it is or some
 bootable media is required for it...as it is in zip
 format...BUT it is a
 .ISO file ...tell me ...what should i do...it will
 run automatically when i
 reboot system and first boot device is
 CDROm...thank
 --
 Muhammad Sohail Arham
 U.E.T. Lahore
 Phone No. 0321-4422406
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RE: [Asterisk-Users] TDM400 pinout

2006-01-26 Thread Chris Bagnall

 Hi I'm looking for a pinout for the above.  Note this has 
 what i'd call
 RJ45 sockets (or someone smart can correct me).  I need to 
 plug into BT (rj13?).

Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12
sockets.

I assume with the mention of BT, you're in the UK. The line is on pins 2+5
of the BT connector, which'd usually translate to the 2 inner pins of an
RJ11 connector (pins 2+3). You should find an old modem cable will do the
job fine.

If your TDM400 really does have RJ45 sockets, then you'd expect the line to
be on the middle pins (pins 4+5), similar to a modtap used in structured
cabling environments.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] VOIP Router

2006-01-26 Thread Arek Bekiersz


Hi,


Try one of Venus 2804, 2808 or 2832 from Tainet corporation.
They support SIP or MGCP and they come with VPN.

http://www.tainet.net
Proceed to Product/VoIP/Venus

--
Regards,
Arek Bekiersz



Mohamed Farid wrote:

Dear All :
I need to link my HQ to some Remote Sites - I need a Router which 
supports VOIP , and VPN

Also the Router Should has 3 FXS ports and 1 FXO ...
The call should be routed from the Remote Site to the HQ through a VPN 
tunnel ( 3DES ) ...

Any Advise ?
Mohamed Farid ,,

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Re: [Asterisk-Users] TDM400 pinout

2006-01-26 Thread bails


Chris Bagnall wrote:
Hi I'm looking for a pinout for the above.  Note this has 
what i'd call
RJ45 sockets (or someone smart can correct me).  I need to 
plug into BT (rj13?).



Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12
sockets.

I assume with the mention of BT, you're in the UK. The line is on pins 2+5
of the BT connector, which'd usually translate to the 2 inner pins of an
RJ11 connector (pins 2+3). You should find an old modem cable will do the
job fine.

If your TDM400 really does have RJ45 sockets, then you'd expect the line to
be on the middle pins (pins 4+5), similar to a modtap used in structured
cabling environments.

Regards,

Chris

Thanks, yes they are rj45, we have had rj12 in he past I look at the above.

Like I said though, pity Digium dont supply the information on there 
site or with the cards, its a bit like everything in life today.  We are 
only the customer, but  we're expected to do the running around.


Bails
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Re: [Asterisk-Users] using sangoma cards as a timesource?

2006-01-26 Thread Matt Florell

Short answer: Yes

Long answer: They use the zaptel drivers and are recognized as a
Zaptel device. You do have to load and configure the Sangoma wanpipe
drivers first, but in the end it'll function as a timing source just
like a Digium card

MATT---

On 1/26/06, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
 hi

 building a new setup, we want to try using sangoma cards. can these
 be used as time sources the same way as TE410Ps?

 thanks

 roy
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Re: [Asterisk-Users] * point to point t1 solution?

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1


Damon Estep wrote:
 Jean-Michel,
 
 I agree with all of your comments, and would be willing to bet $100 that NO 
 AMOUNT OF GOOGLING will answer this question definitively.

I would almost be willing to take that bet... find your exact
configuration is probably not going to happen... however finding enough
information to piece it together is pretty straight forward...

http://www.tek-tips.com/viewthread.cfm?qid=1082800page=10

This link tells you that someone else has connected asterisk to a T1
interface on a Nortel NorthStar.   (first link for asterisk nortel
northstar).

http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation

Whould give you a start as to how to configure the T1 card (zaptel drivers).

http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf

Would give you a start to configure the zaptel.conf


http://www.voip-info.org/wiki-IAX

Gives you the skinny (no pun intended) on IAX and working with it to set
up the trunk...


So it would be logical to assume that if you can connect to the nortel
and asterisk can talk to asterisk via (iax or sip or anything else) ...
you only need to set up an appropriate dial plan to pass extensions back
and forth.

In /etc/zaptel.conf:
   span=1,1,0,esf,b8zs
   bchan=1-23
   dchan=24

In /etc/asterisk/zapata.conf:
   switchtype=national
   context=from-pbx2
   signalling=pri_cpe
   group=0
   channel = 1-23


[from-iax-trunk]
; yeah i know this wouldn't be recommended...
exten = _X.,1,Dial(ZAP/g0/${EXTEN},20)

[from-pbx2]
; still not recommended...
exten = _X.,1,Dial(IAX/pbx2/${EXTEN},20)


This is not Uncharted Territory this is thinking about something as a
sum of its parts not as if No one else has a solution just like me

 
 After reviewing Adrian Carters very informative response regarding TDMoE I am 
 getting closer to what I need to know (now my googles include asterisk AND 
 TDMoE).
 
 This is CLEARLY uncharted territory, while I'll bet it has been done before, 
 no one took the time to document it.
 

No it isn't... I know plenty of people who have connected legacy systems
to IP.  I am doing it with a Merlin Legend.

 My return to the list if I am successful will be to document the config on 
 the wiki... fair exchange?
 
 And, if along the way I find an EXPERT in this area with REAL WORLD 
 PRODUCTION EXPERIENCE I will gladly pay the fees, but I am not about to shell 
 out anything to pay some know-it-all to educate themselves and provide me a 
 half baked solution that has never been put to the real world test.

If you are LOOKING for REAL WORLD, EXACT RepReSenTAtions  (sorry
couldn't resist) of exactly what you have... you are probably going to
be out of luck.  But consider this:

1.  There is plenty of ducumentation on connecting Legacy Systems to
Asterisk
2.  There is plenty of documentation on connect two asterisk systems to
eachother.



 
 D
 
 
 
 
Damon Estep a écrit :


Jean-Michel,

You missed the entire point - the question is IS ASTERISK CAPABLE OF

EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB
BASED HINTS I MIGHT LOOK AT?  Not WILL YOU DO IT FOR ME?


Yes, I think Asterisk can do what you are trying to achieve. No, I don't
know how to do it, and no I won't do it for you since I've never been in
the situation you're in.

As for web based hints, with some experience I've found that google is
as good as asking the mailing list. If you have no success with the
mailing list, the wiki (voip-info.org) and google, you can try the irc
channel #asterisk on freenode.

If that still doesn't work, shelling out a few hundred bucks for a
consultant to help you do it - and train you in the process - is the
other alternative, and is often a good deal.

I've done it a couple of times myself, and it's awesome how far people
can get you and how hard they try when you recognize the value of their
work with some money (as opposed to just asking nicely).

Cheers,
Jean-Michel.

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RE: [Asterisk-Users] Best FXO hardware for home use

2006-01-26 Thread Rich Adamson

I didn't right those products off and in fact use both on a regular
basis. For the price, both are pretty good. However, for a higher price
there are products on the market that _do_ handle echo cancellation in
a very solid fashion (eg, Mediatrix 1204 as one example) regardless of
the analog cable lengths, etc. 

To blatantly suggest the spa3k or TDM are the _recommended_ choices
totally ignores their weaknesses. In other words, chose the product that
fits the situation, and for this list there seems to be a large number
of people that don't have a clue how analog lines are even constructed
let alone measured. There is no such thing as one product that fits all 
needs.



 There is no doubt that given a particular scenario, anything won't work
 properly. This is not necessarily a problem with the SPA3000 or the TDM
 cards, this is much more of a phone line issue. Granted, those devices don't
 handle line issues as well as some other devices (such as the long loop
 issue you mentioned) but to write them off as being poor products I felt was
 a bit overkill. I have several very successful installs with TDM cards and
 SPA3000s and on the other hand I have an install that nothing seems to want
 to work with the PSTN lines that are there. So while I do agree with you on
 what the actual issue is, I don't think it is 100% fair to write off the
 SPA3000 in all cases.
 
 Kerry Garrison
 Director of Technical Services
 Tech Data Pros - Orange County's Mobile IT Service Provider
 (949) 502-7819 x200 - [EMAIL PROTECTED]
 http://www.techdatapros.com
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Rich Adamson
  Sent: Wednesday, January 25, 2006 4:00 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: RE: [Asterisk-Users] Best FXO hardware for home use
  
  
echo cancellation is pretty limited on these cheap devices. 
the spa3000 manual for example states the AEC is limited to 8ms. 
good AECs will handle 64ms or more. in my experience the spa3000 
echo canceller is cranky. it works most but not all of the time.
   
   I have been using one for 6 months without any problems. 
  Make sure you 
   have the most current firmware on it and it should work just fine.
  
  Kerry,
  
  There is an issue with the spa3k (as well as the TDM04b) in 
  terms of handling echo properly on long pstn loops. You are 
  obviously on a relatively short loop if you've not been 
  exposed to the variable echo cancellation issues.
  
  In other words, long pstn loops basically fall outside the 
  limits of the echo cancellation software as someone else 
  already noted.
  
  
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RE: [Asterisk-Users] * point to point t1 solution?

That would not be a nailed up t1 - signaling at both ends would be via asterisk.

I was trying to determine if there is a way to configure asterisk to emulate a 
ptp t1 passively (no signaling) - essentially providing the same type of end to 
end circuit you would get if you ordered a point to point esf/b8zs t1 from the 
telco.

I know how to set up asterisk to talk to the Nortel (where the Nortel thinks 
asterisk is a telco trunk) - but that is not my goal here.

I want to replace a T1 TIE TRUNK between to Nortel's using Digium t1 interfaces 
and an IP link between asterisk boxes, but still allow the Nortel to pass 
signaling directly back and forth.

Still want to take the challenge?


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Sean Cook
 Sent: Thursday, January 26, 2006 6:12 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution?
 
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 
 Damon Estep wrote:
  Jean-Michel,
 
  I agree with all of your comments, and would be willing to bet $100 that
 NO AMOUNT OF GOOGLING will answer this question definitively.
 
 I would almost be willing to take that bet... find your exact
 configuration is probably not going to happen... however finding enough
 information to piece it together is pretty straight forward...
 
 http://www.tek-tips.com/viewthread.cfm?qid=1082800page=10
 
 This link tells you that someone else has connected asterisk to a T1
 interface on a Nortel NorthStar.   (first link for asterisk nortel
 northstar).
 
 http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation
 
 Whould give you a start as to how to configure the T1 card (zaptel
 drivers).
 
 http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf
 
 Would give you a start to configure the zaptel.conf
 
 
 http://www.voip-info.org/wiki-IAX
 
 Gives you the skinny (no pun intended) on IAX and working with it to set
 up the trunk...
 
 
 So it would be logical to assume that if you can connect to the nortel
 and asterisk can talk to asterisk via (iax or sip or anything else) ...
 you only need to set up an appropriate dial plan to pass extensions back
 and forth.
 
 In /etc/zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
 
 In /etc/asterisk/zapata.conf:
switchtype=national
context=from-pbx2
signalling=pri_cpe
group=0
channel = 1-23
 
 
 [from-iax-trunk]
 ; yeah i know this wouldn't be recommended...
 exten = _X.,1,Dial(ZAP/g0/${EXTEN},20)
 
 [from-pbx2]
 ; still not recommended...
 exten = _X.,1,Dial(IAX/pbx2/${EXTEN},20)
 
 
 This is not Uncharted Territory this is thinking about something as a
 sum of its parts not as if No one else has a solution just like me
 
 
  After reviewing Adrian Carters very informative response regarding TDMoE
 I am getting closer to what I need to know (now my googles include
 asterisk AND TDMoE).
 
  This is CLEARLY uncharted territory, while I'll bet it has been done
 before, no one took the time to document it.
 
 
 No it isn't... I know plenty of people who have connected legacy systems
 to IP.  I am doing it with a Merlin Legend.
 
  My return to the list if I am successful will be to document the config
 on the wiki... fair exchange?
 
  And, if along the way I find an EXPERT in this area with REAL WORLD
 PRODUCTION EXPERIENCE I will gladly pay the fees, but I am not about to
 shell out anything to pay some know-it-all to educate themselves and
 provide me a half baked solution that has never been put to the real world
 test.
 
 If you are LOOKING for REAL WORLD, EXACT RepReSenTAtions  (sorry
 couldn't resist) of exactly what you have... you are probably going to
 be out of luck.  But consider this:
 
 1.  There is plenty of ducumentation on connecting Legacy Systems to
 Asterisk
 2.  There is plenty of documentation on connect two asterisk systems to
 eachother.
 
 
 
 
  D
 
 
 
 
 Damon Estep a écrit :
 
 
 Jean-Michel,
 
 You missed the entire point - the question is IS ASTERISK CAPABLE OF
 
 EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY
 WEB
 BASED HINTS I MIGHT LOOK AT?  Not WILL YOU DO IT FOR ME?
 
 
 Yes, I think Asterisk can do what you are trying to achieve. No, I don't
 know how to do it, and no I won't do it for you since I've never been in
 the situation you're in.
 
 As for web based hints, with some experience I've found that google is
 as good as asking the mailing list. If you have no success with the
 mailing list, the wiki (voip-info.org) and google, you can try the irc
 channel #asterisk on freenode.
 
 If that still doesn't work, shelling out a few hundred bucks for a
 consultant to help you do it - and train you in the process - is the
 other alternative, and is often a good deal.
 
 I've done it a couple of times myself, and it's awesome how far people
 can get you and how hard they try when you recognize the value of

Re: [Asterisk-Users] * point to point t1 solution?

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

http://www.voip-info.org/wiki-Asterisk+TDMoE

Damon Estep wrote:
 That would not be a nailed up t1 - signaling at both ends would be via 
 asterisk.
 
 I was trying to determine if there is a way to configure asterisk to emulate 
 a ptp t1 passively (no signaling) - essentially providing the same type of 
 end to end circuit you would get if you ordered a point to point esf/b8zs t1 
 from the telco.
 
 I know how to set up asterisk to talk to the Nortel (where the Nortel thinks 
 asterisk is a telco trunk) - but that is not my goal here.
 
 I want to replace a T1 TIE TRUNK between to Nortel's using Digium t1 
 interfaces and an IP link between asterisk boxes, but still allow the Nortel 
 to pass signaling directly back and forth.
 
 Still want to take the challenge?
 
 
 
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sean Cook
Sent: Thursday, January 26, 2006 6:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * point to point t1 solution?

 
 Damon Estep wrote:
 
Jean-Michel,
 
I agree with all of your comments, and would be willing to bet $100 that
 
 NO AMOUNT OF GOOGLING will answer this question definitively.
 
 I would almost be willing to take that bet... find your exact
 configuration is probably not going to happen... however finding enough
 information to piece it together is pretty straight forward...
 
 http://www.tek-tips.com/viewthread.cfm?qid=1082800page=10
 
 This link tells you that someone else has connected asterisk to a T1
 interface on a Nortel NorthStar.   (first link for asterisk nortel
 northstar).
 
 http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation
 
 Whould give you a start as to how to configure the T1 card (zaptel
 drivers).
 
 http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf
 
 Would give you a start to configure the zaptel.conf
 
 
 http://www.voip-info.org/wiki-IAX
 
 Gives you the skinny (no pun intended) on IAX and working with it to set
 up the trunk...
 
 
 So it would be logical to assume that if you can connect to the nortel
 and asterisk can talk to asterisk via (iax or sip or anything else) ...
 you only need to set up an appropriate dial plan to pass extensions back
 and forth.
 
 In /etc/zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
 
 In /etc/asterisk/zapata.conf:
switchtype=national
context=from-pbx2
signalling=pri_cpe
group=0
channel = 1-23
 
 
 [from-iax-trunk]
 ; yeah i know this wouldn't be recommended...
 exten = _X.,1,Dial(ZAP/g0/${EXTEN},20)
 
 [from-pbx2]
 ; still not recommended...
 exten = _X.,1,Dial(IAX/pbx2/${EXTEN},20)
 
 
 This is not Uncharted Territory this is thinking about something as a
 sum of its parts not as if No one else has a solution just like me
 
 
After reviewing Adrian Carters very informative response regarding TDMoE
 
 I am getting closer to what I need to know (now my googles include
 asterisk AND TDMoE).
 
This is CLEARLY uncharted territory, while I'll bet it has been done
 
 before, no one took the time to document it.
 
 No it isn't... I know plenty of people who have connected legacy systems
 to IP.  I am doing it with a Merlin Legend.
 
 
My return to the list if I am successful will be to document the config
 
 on the wiki... fair exchange?
 
And, if along the way I find an EXPERT in this area with REAL WORLD
 
 PRODUCTION EXPERIENCE I will gladly pay the fees, but I am not about to
 shell out anything to pay some know-it-all to educate themselves and
 provide me a half baked solution that has never been put to the real world
 test.
 
 If you are LOOKING for REAL WORLD, EXACT RepReSenTAtions  (sorry
 couldn't resist) of exactly what you have... you are probably going to
 be out of luck.  But consider this:
 
 1.  There is plenty of ducumentation on connecting Legacy Systems to
 Asterisk
 2.  There is plenty of documentation on connect two asterisk systems to
 eachother.
 
 
 
 
D
 
 
 
 
 
Damon Estep a écrit :
 
 
 
Jean-Michel,

You missed the entire point - the question is IS ASTERISK CAPABLE OF
 
EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY
 
 WEB
 
BASED HINTS I MIGHT LOOK AT?  Not WILL YOU DO IT FOR ME?
 
 
Yes, I think Asterisk can do what you are trying to achieve. No, I don't
know how to do it, and no I won't do it for you since I've never been in
the situation you're in.
 
As for web based hints, with some experience I've found that google is
as good as asking the mailing list. If you have no success with the
mailing list, the wiki (voip-info.org) and google, you can try the irc
channel #asterisk on freenode.
 
If that still doesn't work, shelling out a few hundred bucks for a
consultant to help you do it - and train you in the process - is the
other alternative, and is often a good deal.
 
I've done it a couple of times myself, and it's awesome how far people
can

RE: [Asterisk-Users] * point to point t1 solution?

TDMoE would allow a T1 like connection only over the local Ethernet segment, 
since it is not an IP technology it can not be router across ip networks.

This would be useful to connect 2 asterisk boxes on the same Ethernet segment 
(or with a crossover cable).

The advantage would be lower latency than SIP or IAX - the disadvantage being a 
constant ~1000 packet per second Ethernet flow requires to keep the channels up.

Won't work...



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Sean Cook
 Sent: Thursday, January 26, 2006 6:36 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution?
 
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 http://www.voip-info.org/wiki-Asterisk+TDMoE
 
 Damon Estep wrote:
  That would not be a nailed up t1 - signaling at both ends would be via
 asterisk.
 
  I was trying to determine if there is a way to configure asterisk to
 emulate a ptp t1 passively (no signaling) - essentially providing the same
 type of end to end circuit you would get if you ordered a point to point
 esf/b8zs t1 from the telco.
 
  I know how to set up asterisk to talk to the Nortel (where the Nortel
 thinks asterisk is a telco trunk) - but that is not my goal here.
 
  I want to replace a T1 TIE TRUNK between to Nortel's using Digium t1
 interfaces and an IP link between asterisk boxes, but still allow the
 Nortel to pass signaling directly back and forth.
 
  Still want to take the challenge?
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Sean Cook
 Sent: Thursday, January 26, 2006 6:12 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution?
 
 
  Damon Estep wrote:
 
 Jean-Michel,
 
 I agree with all of your comments, and would be willing to bet $100 that
 
  NO AMOUNT OF GOOGLING will answer this question definitively.
 
  I would almost be willing to take that bet... find your exact
  configuration is probably not going to happen... however finding enough
  information to piece it together is pretty straight forward...
 
  http://www.tek-tips.com/viewthread.cfm?qid=1082800page=10
 
  This link tells you that someone else has connected asterisk to a T1
  interface on a Nortel NorthStar.   (first link for asterisk nortel
  northstar).
 
  http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation
 
  Whould give you a start as to how to configure the T1 card (zaptel
  drivers).
 
  http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf
 
  Would give you a start to configure the zaptel.conf
 
 
  http://www.voip-info.org/wiki-IAX
 
  Gives you the skinny (no pun intended) on IAX and working with it to set
  up the trunk...
 
 
  So it would be logical to assume that if you can connect to the nortel
  and asterisk can talk to asterisk via (iax or sip or anything else) ...
  you only need to set up an appropriate dial plan to pass extensions back
  and forth.
 
  In /etc/zaptel.conf:
 span=1,1,0,esf,b8zs
 bchan=1-23
 dchan=24
 
  In /etc/asterisk/zapata.conf:
 switchtype=national
 context=from-pbx2
 signalling=pri_cpe
 group=0
 channel = 1-23
 
 
  [from-iax-trunk]
  ; yeah i know this wouldn't be recommended...
  exten = _X.,1,Dial(ZAP/g0/${EXTEN},20)
 
  [from-pbx2]
  ; still not recommended...
  exten = _X.,1,Dial(IAX/pbx2/${EXTEN},20)
 
 
  This is not Uncharted Territory this is thinking about something as a
  sum of its parts not as if No one else has a solution just like me
 
 
 After reviewing Adrian Carters very informative response regarding TDMoE
 
  I am getting closer to what I need to know (now my googles include
  asterisk AND TDMoE).
 
 This is CLEARLY uncharted territory, while I'll bet it has been done
 
  before, no one took the time to document it.
 
  No it isn't... I know plenty of people who have connected legacy systems
  to IP.  I am doing it with a Merlin Legend.
 
 
 My return to the list if I am successful will be to document the config
 
  on the wiki... fair exchange?
 
 And, if along the way I find an EXPERT in this area with REAL WORLD
 
  PRODUCTION EXPERIENCE I will gladly pay the fees, but I am not about to
  shell out anything to pay some know-it-all to educate themselves and
  provide me a half baked solution that has never been put to the real
 world
  test.
 
  If you are LOOKING for REAL WORLD, EXACT RepReSenTAtions  (sorry
  couldn't resist) of exactly what you have... you are probably going to
  be out of luck.  But consider this:
 
  1.  There is plenty of ducumentation on connecting Legacy Systems to
  Asterisk
  2.  There is plenty of documentation on connect two asterisk systems to
  eachother.
 
 
 
 
 D
 
 
 
 
 
 Damon Estep a écrit :
 
 
 
 Jean-Michel,
 
 You missed the entire point - the question is IS ASTERISK CAPABLE OF
 
 EMULATING

Re: [Asterisk-Users] * point to point t1 solution?

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Ok... lets get into the network setup... what about bridging a vlan
across your wireless network and sticking both asterisk on the same
segment?   l2tp... (can a forgo the posting of the google links?)  :)



Damon Estep wrote:
 TDMoE would allow a T1 like connection only over the local Ethernet segment, 
 since it is not an IP technology it can not be router across ip networks.
 
 This would be useful to connect 2 asterisk boxes on the same Ethernet segment 
 (or with a crossover cable).
 
 The advantage would be lower latency than SIP or IAX - the disadvantage being 
 a constant ~1000 packet per second Ethernet flow requires to keep the 
 channels up.
 
 Won't work...
 
 
 
 
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sean Cook
Sent: Thursday, January 26, 2006 6:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * point to point t1 solution?

 http://www.voip-info.org/wiki-Asterisk+TDMoE
 
 Damon Estep wrote:
 
That would not be a nailed up t1 - signaling at both ends would be via
 
 asterisk.
 
I was trying to determine if there is a way to configure asterisk to
 
 emulate a ptp t1 passively (no signaling) - essentially providing the same
 type of end to end circuit you would get if you ordered a point to point
 esf/b8zs t1 from the telco.
 
I know how to set up asterisk to talk to the Nortel (where the Nortel
 
 thinks asterisk is a telco trunk) - but that is not my goal here.
 
I want to replace a T1 TIE TRUNK between to Nortel's using Digium t1
 
 interfaces and an IP link between asterisk boxes, but still allow the
 Nortel to pass signaling directly back and forth.
 
Still want to take the challenge?
 
 
 
 
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sean Cook
Sent: Thursday, January 26, 2006 6:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * point to point t1 solution?
 
 
Damon Estep wrote:
 
 
Jean-Michel,
 
I agree with all of your comments, and would be willing to bet $100 that
 
NO AMOUNT OF GOOGLING will answer this question definitively.
 
I would almost be willing to take that bet... find your exact
configuration is probably not going to happen... however finding enough
information to piece it together is pretty straight forward...
 
http://www.tek-tips.com/viewthread.cfm?qid=1082800page=10
 
This link tells you that someone else has connected asterisk to a T1
interface on a Nortel NorthStar.   (first link for asterisk nortel
northstar).
 
http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation
 
Whould give you a start as to how to configure the T1 card (zaptel
drivers).
 
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf
 
Would give you a start to configure the zaptel.conf
 
 
http://www.voip-info.org/wiki-IAX
 
Gives you the skinny (no pun intended) on IAX and working with it to set
up the trunk...
 
 
So it would be logical to assume that if you can connect to the nortel
and asterisk can talk to asterisk via (iax or sip or anything else) ...
you only need to set up an appropriate dial plan to pass extensions back
and forth.
 
In /etc/zaptel.conf:
   span=1,1,0,esf,b8zs
   bchan=1-23
   dchan=24
 
In /etc/asterisk/zapata.conf:
   switchtype=national
   context=from-pbx2
   signalling=pri_cpe
   group=0
   channel = 1-23
 
 
[from-iax-trunk]
; yeah i know this wouldn't be recommended...
exten = _X.,1,Dial(ZAP/g0/${EXTEN},20)
 
[from-pbx2]
; still not recommended...
exten = _X.,1,Dial(IAX/pbx2/${EXTEN},20)
 
 
This is not Uncharted Territory this is thinking about something as a
sum of its parts not as if No one else has a solution just like me
 
 
 
After reviewing Adrian Carters very informative response regarding TDMoE
 
I am getting closer to what I need to know (now my googles include
asterisk AND TDMoE).
 
 
This is CLEARLY uncharted territory, while I'll bet it has been done
 
before, no one took the time to document it.
 
No it isn't... I know plenty of people who have connected legacy systems
to IP.  I am doing it with a Merlin Legend.
 
 
 
My return to the list if I am successful will be to document the config
 
on the wiki... fair exchange?
 
 
And, if along the way I find an EXPERT in this area with REAL WORLD
 
PRODUCTION EXPERIENCE I will gladly pay the fees, but I am not about to
shell out anything to pay some know-it-all to educate themselves and
provide me a half baked solution that has never been put to the real
 
 world
 
test.
 
If you are LOOKING for REAL WORLD, EXACT RepReSenTAtions  (sorry
couldn't resist) of exactly what you have... you are probably going to
be out of luck.  But consider this:
 
1.  There is plenty of ducumentation on connecting Legacy Systems to
Asterisk
2.  There is plenty of documentation on connect two asterisk systems to
eachother.
 
 
 
 
 
D

Re: [Asterisk-Users] * point to point t1 solution?


Damon Estep a écrit :


TDMoE would allow a T1 like connection only over the local Ethernet segment, 
since it is not an IP technology it can not be router across ip networks.
 



You could use OpenVPN to create a virtual tap0 interface over IP, and 
bridge that with your current ethX network.


Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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Re: [Asterisk-Users] Linksys SPA-941 multiple line appearences

2006-01-26 Thread Wilson Pickett

 Has anyone had any experience with the Linksys SPA-941 when it comes to
 multiple line appearences?

This is what the 841 manual says: (maybe the 941 is different?)

The SPA-841 does not support multiple calls on the same Line key. The
corresponding Line key blinks quickly in red on any incoming call. If
there is no other active calls, the SPA-841 will ring with either the
default ring of that  extension or the distinctive ring associated
with the caller. If there is another active call, however, the SPA-841
will not ring the phone, but plays the call-waiting tone to alert the
user.  The SPA-841 supports multiple call-waiting.  In fact, all 4
call appearances can ring at the same time.

Or was that not the question? :)

I find no settings (phone or web interface) on the SPA-941 about call waiting.
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RE: [Asterisk-Users] * point to point t1 solution?

Lets put the TDMoE aside for a minute...

The same trunking could be achieved with SIP or IAX, could it not (with higher 
latency)?

The rest of the question remains - is there a way to get asterisk to output, 
bit for bit, on a t1 interface, the same data that is input on a remote 
asterisk box t1 interface - using any trunking protocol.

This is what would be required to truly emulate a signaling un-aware point to 
point t1 like one that you would get from a telco if you ordered a point to 
point esf/b8zs t1 from A location to Z location.

Pure circuit emulation - not ISDN/CAS/EM signaled voice.

Does that clarify the question at all?



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
 Sent: Thursday, January 26, 2006 6:53 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution?
 
 Damon Estep a écrit :
 
 TDMoE would allow a T1 like connection only over the local Ethernet
 segment, since it is not an IP technology it can not be router across ip
 networks.
 
 
 
 You could use OpenVPN to create a virtual tap0 interface over IP, and
 bridge that with your current ethX network.
 
 Cheers,
 Jean-Michel.
 
 --
 Jean-Michel Hiver - http://ykoz.net/
 Découvrez la Réunion des Technologies IP  Telecom
 TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
 
 
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RE: [Asterisk-Users] * point to point t1 solution?

1000pps TDMoE plus vlan tagging, plus l2tp over 10mbps microwave?

I assume you have not tried this before, correct?

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Sean Cook
 Sent: Thursday, January 26, 2006 6:47 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution?

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Ok... lets get into the network setup... what about bridging a vlan
 across your wireless network and sticking both asterisk on the same
 segment?   l2tp... (can a forgo the posting of the google links?)  :)

 Damon Estep wrote:
  TDMoE would allow a T1 like connection only over the local Ethernet
 segment, since it is not an IP technology it can not be router across ip
 networks.

  This would be useful to connect 2 asterisk boxes on the same Ethernet
 segment (or with a crossover cable).

  The advantage would be lower latency than SIP or IAX - the disadvantage
 being a constant ~1000 packet per second Ethernet flow requires to keep
 the channels up.

  Won't work...

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Sean Cook
 Sent: Thursday, January 26, 2006 6:36 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution?

  http://www.voip-info.org/wiki-Asterisk+TDMoE

  Damon Estep wrote:

 That would not be a nailed up t1 - signaling at both ends would be via

  asterisk.

 I was trying to determine if there is a way to configure asterisk to

  emulate a ptp t1 passively (no signaling) - essentially providing the
 same
  type of end to end circuit you would get if you ordered a point to point
  esf/b8zs t1 from the telco.

 I know how to set up asterisk to talk to the Nortel (where the Nortel

  thinks asterisk is a telco trunk) - but that is not my goal here.

 I want to replace a T1 TIE TRUNK between to Nortel's using Digium t1

  interfaces and an IP link between asterisk boxes, but still allow the
  Nortel to pass signaling directly back and forth.

 Still want to take the challenge?

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Sean Cook
 Sent: Thursday, January 26, 2006 6:12 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution?

 Damon Estep wrote:

 Jean-Michel,

 I agree with all of your comments, and would be willing to bet $100
 that

 NO AMOUNT OF GOOGLING will answer this question definitively.

 I would almost be willing to take that bet... find your exact
 configuration is probably not going to happen... however finding enough
 information to piece it together is pretty straight forward...

 http://www.tek-tips.com/viewthread.cfm?qid=1082800page=10

 This link tells you that someone else has connected asterisk to a T1
 interface on a Nortel NorthStar.   (first link for asterisk nortel
 northstar).

 http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation

 Whould give you a start as to how to configure the T1 card (zaptel
 drivers).

 http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf

 Would give you a start to configure the zaptel.conf

 http://www.voip-info.org/wiki-IAX

 Gives you the skinny (no pun intended) on IAX and working with it to set
 up the trunk...

 So it would be logical to assume that if you can connect to the nortel
 and asterisk can talk to asterisk via (iax or sip or anything else) ...
 you only need to set up an appropriate dial plan to pass extensions back
 and forth.

 In /etc/zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

 In /etc/asterisk/zapata.conf:
switchtype=national
context=from-pbx2
signalling=pri_cpe
group=0
channel = 1-23

 [from-iax-trunk]
 ; yeah i know this wouldn't be recommended...
 exten = _X.,1,Dial(ZAP/g0/${EXTEN},20)

 [from-pbx2]
 ; still not recommended...
 exten = _X.,1,Dial(IAX/pbx2/${EXTEN},20)

 This is not Uncharted Territory this is thinking about something as a
 sum of its parts not as if No one else has a solution just like me

 After reviewing Adrian Carters very informative response regarding
 TDMoE

 I am getting closer to what I need to know (now my googles include
 asterisk AND TDMoE).

 This is CLEARLY uncharted territory, while I'll bet it has been done

 before, no one took the time to document it.

 No it isn't... I know plenty of people who have connected legacy systems
 to IP.  I am doing it with a Merlin Legend.

 My return to the list if I am successful will be to document the config

 on the wiki... fair exchange?

 And, if along the way I find an EXPERT in this area with REAL WORLD

 PRODUCTION EXPERIENCE I will gladly pay the fees, but I am not about to
 shell out

RE: [Asterisk-Users] using sangoma cards as a timesource?

And in some (many) cases it will do so while sharing an interrupt with a
NIC and disk controller!

We run sangoma a104 cards in Dell SC1425 1U servers with great success
under heavy load.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matt Florell
 Sent: Thursday, January 26, 2006 5:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] using sangoma cards as a timesource?
 
 Short answer: Yes
 
 Long answer: They use the zaptel drivers and are recognized as a
 Zaptel device. You do have to load and configure the Sangoma wanpipe
 drivers first, but in the end it'll function as a timing source just
 like a Digium card
 
 MATT---
 
 On 1/26/06, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
  hi
 
  building a new setup, we want to try using sangoma cards. can these
  be used as time sources the same way as TE410Ps?
 
  thanks
 
  roy
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Re: [Asterisk-Users] * point to point t1 solution?


Damon Estep a écrit :


Lets put the TDMoE aside for a minute...

The same trunking could be achieved with SIP or IAX, could it not (with higher 
latency)?

The rest of the question remains - is there a way to get asterisk to output, 
bit for bit, on a t1 interface, the same data that is input on a remote 
asterisk box t1 interface - using any trunking protocol.

This is what would be required to truly emulate a signaling un-aware point to 
point t1 like one that you would get from a telco if you ordered a point to point 
esf/b8zs t1 from A location to Z location.

Pure circuit emulation - not ISDN/CAS/EM signaled voice.
 

You might want to take a look at rad.com array of products. They sell 
small boxes which cost around €3k each and which can do exactly what you 
are looking after, which they call TDMoIP. Of course, this is getter 
further away from Asterisk :(


Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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Re: [Asterisk-Users] Fast AGI Options. Eeeek!

2006-01-26 Thread Simone Cittadini


Sig Lange ha scritto:



 I have successfully written FastAGI applications in python, and it 
was a good experience.


Do you have some template code you can share ? or references to point us 
to ?

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Re: [Asterisk-Users] * point to point t1 solution?

2006-01-26 Thread Patrick Conroy

Damon,Unless I misunderstand what you are looking for, a P2P T1 would be handled by the kernel, not by asterisk. If you want to use digium cards, you would still need zaptel, or you could use a sangoma card on each end and their wanrouter drivers. Asterisk would obviously be involved in the SIP or IAX connection to pass calls, but not with the P2P T1. This page may help:
http://voip-info.org/wiki/view/Asterisk+Data+ConfigurationThis is based on a T1 using Cisco HDLC, but I have done the same thing with PPP.
Hope that helps,PatrickOn 1/26/06, Damon Estep [EMAIL PROTECTED] wrote:
Lets put the TDMoE aside for a minute...The same trunking could be achieved with SIP or IAX, could it not (with higher latency)?
The rest of the question remains - is there a way to get asterisk to output, bit for bit, on a t1 interface, the same data that is input on a remote asterisk box t1 interface - using any trunking protocol.
This is what would be required to truly emulate a signaling un-aware point to point t1 like one that you would get from a telco if you ordered a point to point esf/b8zs t1 from A location to Z location.
Pure circuit emulation - not ISDN/CAS/EM signaled voice.Does that clarify the question at all? -Original Message- From: 
[EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Jean-Michel Hiver Sent: Thursday, January 26, 2006 6:53 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * point to point t1 solution? Damon Estep a écrit : TDMoE would allow a T1 like connection only over the local Ethernet
 segment, since it is not an IP technology it can not be router across ip networks.   You could use OpenVPN to create a virtual tap0 interface over IP, and bridge that with your current ethX network.
 Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP  Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
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Re: [Asterisk-Users] TDM400 pinout

2006-01-26 Thread BJ Weschke

On 1/26/06, bails [EMAIL PROTECTED] wrote:
 Chris Bagnall wrote:
 Hi I'm looking for a pinout for the above.  Note this has
 what i'd call
 RJ45 sockets (or someone smart can correct me).  I need to
 plug into BT (rj13?).
 
 
  Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12
  sockets.
 
  I assume with the mention of BT, you're in the UK. The line is on pins 2+5
  of the BT connector, which'd usually translate to the 2 inner pins of an
  RJ11 connector (pins 2+3). You should find an old modem cable will do the
  job fine.
 
  If your TDM400 really does have RJ45 sockets, then you'd expect the line to
  be on the middle pins (pins 4+5), similar to a modtap used in structured
  cabling environments.
 
  Regards,
 
  Chris
 Thanks, yes they are rj45, we have had rj12 in he past I look at the above.

 Like I said though, pity Digium dont supply the information on there
 site or with the cards, its a bit like everything in life today.  We are
 only the customer, but  we're expected to do the running around.


 Earlier versions of the TDM400 I believe were RJ45. They were changed
to RJ11 I think I had heard at one point for compliance with some
telco standards outside the US. But, in either case, yes, the middle
pair is the active pair for your FXO/FXS ports on these cards
whether RJ11 or RJ45.

--
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http://www.btwtech.com/
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Re: [Asterisk-Users] asterisk 1.2.3 call problem

2006-01-26 Thread Matt Riddell (IT)

What's in:

#include iax_additional.conf
#include iax_custom.conf

-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] * point to point t1 solution?

2006-01-26 Thread Matt Riddell (IT)

Damon Estep wrote:
 I agree with all of your comments, and would be willing to bet $100 that NO 
 AMOUNT OF GOOGLING will answer this question definitively.

Um, if you google for pri_net pri_cpi and Asterisk, then I bet it will
return a response to your liking.

-- 
Cheers,

Matt Riddell
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RE: [Asterisk-Users] * point to point t1 solution?

saw those, according to RAD they occupy 2mbps even when idle. about $750/each 
for t1

From: [EMAIL PROTECTED] on behalf of Jean-Michel Hiver
Sent: Thu 1/26/2006 7:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * point to point t1 solution?

Damon Estep a écrit :

Lets put the TDMoE aside for a minute...

The same trunking could be achieved with SIP or IAX, could it not (with higher 
latency)?

The rest of the question remains - is there a way to get asterisk to output, 
bit for bit, on a t1 interface, the same data that is input on a remote 
asterisk box t1 interface - using any trunking protocol.

This is what would be required to truly emulate a signaling un-aware point 
to point t1 like one that you would get from a telco if you ordered a point to 
point esf/b8zs t1 from A location to Z location.

Pure circuit emulation - not ISDN/CAS/EM signaled voice.

You might want to take a look at rad.com array of products. They sell
small boxes which cost around EUR3k each and which can do exactly what you
are looking after, which they call TDMoIP. Of course, this is getter
further away from Asterisk :(

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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RE: [Asterisk-Users] RE: IAX Provider

2006-01-26 Thread Ross C









 customer service sucks
as usual

I 100% agree. I havent been
able to complete a call ever. No response from customer service.

Whatever company can provide reliable
service, great support and a good selection of local numbers without charging
out the butt, will do very well IMO. Too many companies dont spend
enough time on customer service, which ends up driving a LOT of business away
(business that they would have had in the bag, had they responded to customers).
Its really not hard to provide good cust service over the
Internet. It doesnt take very long to reply to an email inquiry or
fix the little problems most people have.

Is it an employee thing; do these
companies just not have enough people to get everything done? Or is it
that the owners just dont really care and they think they can succeed
anyway (w/o good service)? Its really ridiculous.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, January 26, 2006
1:35 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RE:
IAX Provider







Truely will all due respects to all, I guess Kaleb has a tie up with
them.. 











I could never get my sixtel try call out except once. They customer
service sucks as usual











I regret paying to them...Their sample config are all done from my side
my my calls comes out with 'NO ANSWER











Dan







On 25/01/06, Dovid
Bender [EMAIL PROTECTED]
wrote: 

Let me guess you have no affiliation with them what so
ever ? no commision on accounts either ?
--- Kaleb L. Kunzler 
[EMAIL PROTECTED]
wrote:

 I use iax.cc and find their service to be superior
 to ANY other VOIP
 provider I have tried.Their prices are 
 competitive, My calls always go
 through, I always get my calls, I couldn't think of
 a better provider.They
 are picky with the context that you use in your
 IAX.cc, but as long as you 
 use the sample config that they provide it works
 beautifully. You only need
 $5 to open the account, that really isn't bad at
 all, others like
 sellvoip.net (bad) require $25 to open
an account.
 If you need help getting
 their service to work for you, please contact me off
 list.


 On 1/25/06, [EMAIL PROTECTED]
[EMAIL PROTECTED]
 wrote:
 
  You can try with www,iax.cc too but i guesst not
 luck with a test
  account.. 
 
  Dan
 
 
  On 25/01/06, Nilesh Londhe [EMAIL PROTECTED]
 wrote:
  
   I use www.voipjet.com and
find it OK.
  
   On 1/24/06, Roberto Pereyra 
 [EMAIL PROTECTED]
wrote: 
Hi

   
I looking a good IAX service for a emerging
 voip provider.
   
Better with a test account to try. 
   
Thanks in advance.
   
roberto

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Re: [Asterisk-Users] TDM400 pinout

2006-01-26 Thread bails


BJ Weschke wrote:

On 1/26/06, bails [EMAIL PROTECTED] wrote:


Chris Bagnall wrote:


Hi I'm looking for a pinout for the above.  Note this has
what i'd call
RJ45 sockets (or someone smart can correct me).  I need to
plug into BT (rj13?).



Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12
sockets.

I assume with the mention of BT, you're in the UK. The line is on pins 2+5
of the BT connector, which'd usually translate to the 2 inner pins of an
RJ11 connector (pins 2+3). You should find an old modem cable will do the
job fine.

If your TDM400 really does have RJ45 sockets, then you'd expect the line to
be on the middle pins (pins 4+5), similar to a modtap used in structured
cabling environments.

Regards,

Chris


Thanks, yes they are rj45, we have had rj12 in he past I look at the above.

Like I said though, pity Digium dont supply the information on there
site or with the cards, its a bit like everything in life today.  We are
only the customer, but  we're expected to do the running around.




 Earlier versions of the TDM400 I believe were RJ45. They were changed
to RJ11 I think I had heard at one point for compliance with some
telco standards outside the US. But, in either case, yes, the middle
pair is the active pair for your FXO/FXS ports on these cards
whether RJ11 or RJ45.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Thanks I can confirm that this is indeed correct

Bails
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RE: [Asterisk-Users] Bootable CD?




To 
clarify: You have to write it as a DISK IMAGE. If you simply drag the ISO file 
to your Nero project and write it, you will get a CD with a single file on it - 
the ISO image - and not the CONTENTS of the ISO Image. 

1. Run 
Nero
2. In 
the New Compilation dialog click Cancel
3. 
Click File  Burn Image, select" All Files" under Files of Type and pick your 
ISO. 
4. 
Click OK, then click Write

hth



  -Original Message-From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]Sent: Thursday, January 26, 2006 1:31 
  AMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] Bootable 
  CD?
  yup... its a bootable image.. go ahead and just write it 
directly...
  
  
  Dan
  On 26/01/06, Sohail 
  Arham [EMAIL PROTECTED] 
  wrote: 
  ahan...then 
it mean it doesnt need to uncompress it..juss write on cd by nero burning 
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Re: [Asterisk-Users] * point to point t1 solution? / alternatives

2006-01-26 Thread Bill Michaelson

This has been an interesting discussion for me (except for the 
sniping).  The last post led me, out of curiosity, to this wiki entry:


http://www.voip-info.org/wiki-Asterisk+TDMoE

I was unaware of this feature, and it looks pretty good.  I've been 
pondering replacing some T1's by leveraging IP capacity but of course 
have run up against the QoS issue.  My idea was different...


I don't have production experience with precisely this type of 
application, but I ask for validation from this list.  Pardon me for 
stating what is undoubtedly obvious to many...


The key to assuring adequate performance in replacing a TDM link with IP 
is to assure that adequate idle time is reserved for voice on the IP 
segment(s) involved in the route.  In this way, latency can be 
stabilized, and if maintained below a certain (arbitrary) threshold, 
performance can be deemed acceptable.


The first step, of course, is to assure that the virtual TDM allocation 
does not exceed the available IP bandwidth (so leave a margin, which is 
huge in the example given).  The next step is to use routers which 
respect the TOS field (however it is used; diffserv/whatever), and 
finally, to assure that no non-VoIP traffic can be injected into the 
path with higher routing priority.


On a point-to-point link, a pair of typical Linux boxes can do all 
this.  Given the original problem, I would place Asterisk boxes at 
either end of the link, and have them blend the ordinary traffic with 
the VoIP traffic (which would probably use IAX to relay calls between 
the T1s), while assuring (enforcing) that VoIP packets are marked as 
highest priority.  There are varied ways of accomplishing this, and a 
good reference which I've used in the past can be found at:


http://www.lartc.org/lartc.html

Additionally, I think one could use the tunneling  techniques described 
in that guide to encapsulate the non-VoIP traffic such that its packets' 
originally marked TOS values are preserved for transit outside the 
segment used for TDM emulation.  In this way, that part of the segment 
bandwidth not required for VoIP would function as a dedicated link, 
allowing other prioritization of traffic such as interactive vs. bulk 
(or even other voice!), with the added advantage that it could use the 
reserved VoIP bandwidth when it is otherwise not required (albeit in the 
case of a T-1 over 10Mb, that's insignificant).


Is this easier or harder than TDMoE as described?  Does the TDMoE shared 
idle bandwidth?  What about stability (I'm thinking of SW releases)?  
What other drawbacks or advantages are there?



Date: Wed, 25 Jan 2006 23:53:59 -0700
From: Damon Estep [EMAIL PROTECTED]
Subject: [Asterisk-Users] * point to point t1 solution?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com

Can anyone point me to a reference or sample config for bypassing a
nailed up (point to point) t1 between two PBXs with asterisk and a pair
of t1 cards?

Right now I have 2 Nortel norstars connected to each other via a leased
line t1. I also have a solid 10mbps low latency microwave link between
the 2 sites.

My goal is to run an asterisk box at each end with a t1 card and
Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in a
relatively dumb configuration, with the goal of migrating off of the
norstars eventually.

In past situations I would have done this with a pair of Cisco routers
with T1 interfaces in them but in this case I want to get asterisk into
the picture as an eventual replacement for the norstars.


 



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Re: [Asterisk-Users] * point to point t1 solution?

2006-01-26 Thread Bill Michaelson

You've clarified your requirements for me.  Please indulge me - I really 
want to understand - what are the application implications of this?  In 
other words, what system behavioral changes will your users experience 
in the various scenarios (pure circuit emulation vs. relay via IAX or 
similar)?

Date: Thu, 26 Jan 2006 07:00:02 -0700
From: Damon Estep [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] * point to point t1 solution?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain;   charset=iso-8859-1

Lets put the TDMoE aside for a minute...

The same trunking could be achieved with SIP or IAX, could it not (with higher 
latency)?

The rest of the question remains - is there a way to get asterisk to output, 
bit for bit, on a t1 interface, the same data that is input on a remote 
asterisk box t1 interface - using any trunking protocol.

This is what would be required to truly emulate a signaling un-aware point to 
point t1 like one that you would get from a telco if you ordered a point to point 
esf/b8zs t1 from A location to Z location.

Pure circuit emulation - not ISDN/CAS/EM signaled voice.

Does that clarify the question at all?

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RE: [Asterisk-Users] * point to point t1 solution?

2006-01-26 Thread Ross C









Uhh..maybe you should ask
Jean-Michel for a refund.

Wait, you havent paid a dime for
this. Or Asterisk. Or most of the Asterisk add-ons.

I always see people getting mad at other
people for bad advice or bad answers to their
questions; people seem to forget that all this stuff is FREE.  If Jean-Michels
advice isnt what youre looking for, say Thanks for the
info, but Id really like to know..  (geez, I feel like
someones mom).  Hes taken time out of HIS day to try to help YOU
for FREE.

If a high level of support and definitive
answers are a must for your situation, pay someone with experience, or see the
following:



expensive IP telephony

http://www.cisco.com

http://www.nortel.com

http://www.inter-tel.com

http://www.avaya.com

http://www.3com.com

/expensive IP telephony











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, January 26, 2006
3:22 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] *
point to point t1 solution?





Actually, it is a quite appropriate
response to ANYONE that includes this type of comment in their reply



You probably need a couple
of T1 cards, and some paid consulting to get it working (I've never done it
myself but that's how I would do it if I was in a hurry)



Perhaps something like this would have been better received;



I know it can (or cannot) be done, and here is the name of
someone that might be willing to help you for a fee



Look back though the archives and you will see that I have had some
participation here myself in the past



D













From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Simon Woodhead
Sent: Thursday, January 26, 2006
2:01 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] *
point to point t1 solution?





Bad day Damon? I think
your comments are a little harsh towards someone who is an active and informed
contributor to the list. Jean-Michel could have ignored you but he chose to
share what he could. Maybe someone else will have the complete answer to your
question. 



On 1/26/06, Damon
Estep [EMAIL PROTECTED]
wrote:

Jean-Michel,

You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING A
POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I
MIGHT LOOK AT?Not WILL YOU DO IT FOR ME?

Your response to this post was un-informative and quite frankly it is the type
of useless response that most mailing lists and newsgroups could do without.

Damon

 -Original Message-
 From: [EMAIL PROTECTED]
[mailto:asterisk-users-
 [EMAIL PROTECTED] ]
On Behalf Of Jean-Michel Hiver
 Sent: Thursday, January 26, 2006 1:36 AM
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution? 

 Damon Estep a écrit :

  Can anyone point me to a reference or sample config for bypassing a
  nailed up (point to point) t1 between two PBXs with asterisk and a
  pair of t1 cards? 
 
 
 
  Right now I have 2 Nortel norstars connected to each other via a
  leased line t1. I also have a solid 10mbps low latency microwave link
  between the 2 sites. 
 
 You probably need a couple of T1 cards, and some paid consulting to get
 it working (I've never done it myself but that's how I would do it if I
 was in a hurry)


  My goal is to run an asterisk box at each end with a t1 card and 
  Ethernet card to act as a TDMSIP gateway to bypass the nailed
T1 in
  a relatively dumb configuration, with the goal of migrating off of
the
  norstars eventually.
 
 If it's a point to point Asterisk - Asterisk configuration, why
use
 SIP rather than IAX? IAX configuration is very easy, so once you get the
 norstar - asterisk link up it'll be a piece of cake. 

 Cheers,
 Jean-Michel.

 --
 Jean-Michel Hiver - http://ykoz.net/
 Découvrez la Réunion des Technologies IP  Telecom
 TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE 


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RE: [Asterisk-Users] * point to point t1 solution? / alternatives

2006-01-26 Thread Steve Langstaff

Remember, however that TDMoE is TDMoE, not TDMoIP - it's not routable
(unless you encapsulate it somehow, I guess).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bill
Michaelson
Sent: 26 January 2006 14:58
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] * point to point t1 solution? /
alternatives

This has been an interesting discussion for me (except for the 
sniping).  The last post led me, out of curiosity, to this wiki entry:

http://www.voip-info.org/wiki-Asterisk+TDMoE

I was unaware of this feature, and it looks pretty good.  I've been 
pondering replacing some T1's by leveraging IP capacity but of course 
have run up against the QoS issue.  My idea was different...

I don't have production experience with precisely this type of 
application, but I ask for validation from this list.  Pardon me for 
stating what is undoubtedly obvious to many...

The key to assuring adequate performance in replacing a TDM link with IP 
is to assure that adequate idle time is reserved for voice on the IP 
segment(s) involved in the route.  In this way, latency can be 
stabilized, and if maintained below a certain (arbitrary) threshold, 
performance can be deemed acceptable.

The first step, of course, is to assure that the virtual TDM allocation 
does not exceed the available IP bandwidth (so leave a margin, which is 
huge in the example given).  The next step is to use routers which 
respect the TOS field (however it is used; diffserv/whatever), and 
finally, to assure that no non-VoIP traffic can be injected into the 
path with higher routing priority.

On a point-to-point link, a pair of typical Linux boxes can do all 
this.  Given the original problem, I would place Asterisk boxes at 
either end of the link, and have them blend the ordinary traffic with 
the VoIP traffic (which would probably use IAX to relay calls between 
the T1s), while assuring (enforcing) that VoIP packets are marked as 
highest priority.  There are varied ways of accomplishing this, and a 
good reference which I've used in the past can be found at:

http://www.lartc.org/lartc.html

Additionally, I think one could use the tunneling  techniques described 
in that guide to encapsulate the non-VoIP traffic such that its packets' 
originally marked TOS values are preserved for transit outside the 
segment used for TDM emulation.  In this way, that part of the segment 
bandwidth not required for VoIP would function as a dedicated link, 
allowing other prioritization of traffic such as interactive vs. bulk 
(or even other voice!), with the added advantage that it could use the 
reserved VoIP bandwidth when it is otherwise not required (albeit in the 
case of a T-1 over 10Mb, that's insignificant).

Is this easier or harder than TDMoE as described?  Does the TDMoE shared 
idle bandwidth?  What about stability (I'm thinking of SW releases)?  
What other drawbacks or advantages are there?

Date: Wed, 25 Jan 2006 23:53:59 -0700
From: Damon Estep [EMAIL PROTECTED]
Subject: [Asterisk-Users] * point to point t1 solution?
To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com

Can anyone point me to a reference or sample config for bypassing a
nailed up (point to point) t1 between two PBXs with asterisk and a pair
of t1 cards?

Right now I have 2 Nortel norstars connected to each other via a leased
line t1. I also have a solid 10mbps low latency microwave link between
the 2 sites.

My goal is to run an asterisk box at each end with a t1 card and
Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in a
relatively dumb configuration, with the goal of migrating off of the
norstars eventually.

 In past situations I would have done this with a pair of Cisco routers
with T1 interfaces in them but in this case I want to get asterisk into
the picture as an eventual replacement for the norstars.

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Re: [Asterisk-Users] * point to point t1 solution?

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Ross,

I was a little frustrated with Damon's initial reaction to the post as
well.  However, we have moved past this ... This is actually turning out
to be quite an interesting thread, lets not get side-tract.

Regards,

Sean

Ross C wrote:
 Uhh?..maybe you should ask Jean-Michel for a refund.
 
 Wait, you haven?t paid a dime for this. Or Asterisk. Or most of the
 Asterisk add-ons.
 
 I always see people getting mad at other people for ?bad advice? or ?bad
 answers? to their questions; people seem to forget that all this stuff
 is FREE.  If Jean-Michel?s advice isn?t what you?re looking for, say
 ?Thanks for the info, but I?d really like to know?..?  (geez, I feel
 like someone?s mom).  He?s taken time out of HIS day to try to help YOU
 for FREE.
 
 If a high level of support and definitive answers are a must for your
 situation, pay someone with experience, or see the following:
 
  
 
 expensive IP telephony
 
 http://www.cisco.com http://www.cisco.com/
 
 http://www.nortel.com http://www.nortel.com/
 
 http://www.inter-tel.com http://www.inter-tel.com/
 
 http://www.avaya.com http://www.avaya.com/
 
 http://www.3com.com http://www.3com.com/
 
 /expensive IP telephony
 
  
 
 
 
 *From:* [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] *On Behalf Of *Damon Estep
 *Sent:* Thursday, January 26, 2006 3:22 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* RE: [Asterisk-Users] * point to point t1 solution?
 
  
 
 Actually, it is a quite appropriate response to ANYONE that includes
 this type of comment in their reply
 
  
 
 ?You probably need a couple of T1 cards, and some paid consulting to get
 it working (I've never done it myself but that's how I would do it if I
 was in a hurry)?
 
  
 
 Perhaps something like this would have been better received;
 
  
 
 ?I know it can (or cannot) be done, and here is the name of someone that
 might be willing to help you for a fee?
 
  
 
 Look back though the archives and you will see that I have had some
 participation here myself  in the past?
 
  
 
 D
 
  
 
 
 
 *From:* [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] *On Behalf Of *Simon
 Woodhead
 *Sent:* Thursday, January 26, 2006 2:01 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [Asterisk-Users] * point to point t1 solution?
 
  
 
 Bad day Damon? I think your comments are a little harsh towards someone
 who is an active and informed contributor to the list. Jean-Michel could
 have ignored you but he chose to share what he could. Maybe someone else
 will have the complete answer to your question.
 
 On 1/26/06, *Damon Estep* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:
 
 Jean-Michel,
 
 You missed the entire point - the question is IS ASTERISK CAPABLE OF
 EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY
 WEB BASED HINTS I MIGHT LOOK AT?  Not WILL YOU DO IT FOR ME?
 
 Your response to this post was un-informative and quite frankly it is
 the type of useless response that most mailing lists and newsgroups
 could do without.
 
 Damon
 
 -Original Message-
 From: [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] [mailto:asterisk-users-
 mailto:asterisk-users-
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf
 Of Jean-Michel Hiver
 Sent: Thursday, January 26, 2006 1:36 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] * point to point t1 solution?

 Damon Estep a écrit :

  Can anyone point me to a reference or sample config for bypassing a
  nailed up (point to point) t1 between two PBXs with asterisk and a
  pair of t1 cards?
 
 
 
  Right now I have 2 Nortel norstars connected to each other via a
  leased line t1. I also have a solid 10mbps low latency microwave link
  between the 2 sites.
 
 You probably need a couple of T1 cards, and some paid consulting to get
 it working (I've never done it myself but that's how I would do it if I
 was in a hurry)


  My goal is to run an asterisk box at each end with a t1 card and
  Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in
  a relatively dumb configuration, with the goal of migrating off of the
  norstars eventually.
 
 If it's a point to point Asterisk - Asterisk configuration, why use
 SIP rather than IAX? IAX configuration is very easy, so once you get the
 norstar - asterisk link up it'll be a piece of cake.

 Cheers,
 Jean-Michel.

 --
 Jean-Michel Hiver - http://ykoz.net/
 Découvrez la Réunion des Technologies IP  Telecom
 TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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[Asterisk-Users] Snom360 Sidecar Asterisk

2006-01-26 Thread c waddy

We are looking to replace our existing Legacy PBX with Asterisk. Our receptionist currently has a light display for a certain extension when someone is on a call. When she needs to transfer she simply hits that button.


Is it possible to use a snom360 + Sidecar to monitor 30 extensions and make transfers using the buttons? Does the Cisco 7960 expansion module work with asterisk?

Thanks.
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Re: [Asterisk-Users] * point to point t1 solution? / alternatives

2006-01-26 Thread Bill Michaelson

Right - so I will assume this makes it slightly more efficient in that 
respect.  And of course, any solution that uses multiple hops brings in 
a raft of considerations for limiting interference by other data streams 
- the essential QoS question.



Date: Thu, 26 Jan 2006 15:16:25 -
From: Steve Langstaff [EMAIL PROTECTED]

Remember, however that TDMoE is TDMoE, not TDMoIP - it's not routable
(unless you encapsulate it somehow, I guess).

 



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RE: [Asterisk-Users] * point to point t1 solution?

Thanks Matt,
 
PRI signalling means that calls and answered and dialed (aka signalled) by 
asterisk, the goal is to maintain the signalling between the two nortel boxes.
 
I have gathered that raw point to point circuit emulation is not possible on 
asterisk...
 
I am aware of how to connect a PBX to asterisk using ISDN PRI signalling.



From: [EMAIL PROTECTED] on behalf of Matt Riddell (IT)
Sent: Thu 1/26/2006 7:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * point to point t1 solution?



Damon Estep wrote:
 I agree with all of your comments, and would be willing to bet $100 that NO 
 AMOUNT OF GOOGLING will answer this question definitively.

Um, if you google for pri_net pri_cpi and Asterisk, then I bet it will
return a response to your liking.

--
Cheers,

Matt Riddell
___

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RE: [Asterisk-Users] * point to point t1 solution?

gladly,
 
circuit emulation will;
 
1. eliminate the need to reconfigure the exisitng hardware.
2. improve the chances that fax and analog modem devices will still work.
3. NOT change any dialing patterns or extensons numbering.
 
there are other, but they are less significant
 



From: [EMAIL PROTECTED] on behalf of Bill Michaelson
Sent: Thu 1/26/2006 8:08 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] * point to point t1 solution?



You've clarified your requirements for me.  Please indulge me - I really
want to understand - what are the application implications of this?  In
other words, what system behavioral changes will your users experience
in the various scenarios (pure circuit emulation vs. relay via IAX or
similar)?


Date: Thu, 26 Jan 2006 07:00:02 -0700
From: Damon Estep [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] * point to point t1 solution?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain;   charset=iso-8859-1

Lets put the TDMoE aside for a minute...

The same trunking could be achieved with SIP or IAX, could it not (with higher 
latency)?

The rest of the question remains - is there a way to get asterisk to output, 
bit for bit, on a t1 interface, the same data that is input on a remote 
asterisk box t1 interface - using any trunking protocol.

This is what would be required to truly emulate a signaling un-aware point to 
point t1 like one that you would get from a telco if you ordered a point to 
point esf/b8zs t1 from A location to Z location.

Pure circuit emulation - not ISDN/CAS/EM signaled voice.

Does that clarify the question at all?



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RE: [Asterisk-Users] * point to point t1 solution? / alternatives

I've seen this discussion before. The conclusion was, it is possible to
route TDMoE through a VPN tunnel depending on the tunnel setup you are using
(bridge + tunnel for example) however the latency would make it useless.
TDMoE is designed for the same network. Unfortuanely I can't find a link for
it, but I remember it distinctly. 

Another, large issue, is that TDMoE uses T1 - style bandwidth constantly
whether it is in use or not. Even if it were possible to route it, and even
if the latency problem was solved, can you imagine your bandwidth surcharge
of ~1.5Mbps constant? 

At the end of the day, emulating TDM through the use of IAX and a well
written dialplan is totally the way to go. 

-Original Message-
From: Steve Langstaff [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 26, 2006 8:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] * point to point t1 solution? /
alternatives


Remember, however that TDMoE is TDMoE, not TDMoIP - it's not routable
(unless you encapsulate it somehow, I guess).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bill
Michaelson
Sent: 26 January 2006 14:58
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] * point to point t1 solution? /
alternatives


This has been an interesting discussion for me (except for the 
sniping).  The last post led me, out of curiosity, to this wiki entry:

http://www.voip-info.org/wiki-Asterisk+TDMoE

I was unaware of this feature, and it looks pretty good.  I've been 
pondering replacing some T1's by leveraging IP capacity but of course 
have run up against the QoS issue.  My idea was different...

I don't have production experience with precisely this type of 
application, but I ask for validation from this list.  Pardon me for 
stating what is undoubtedly obvious to many...

The key to assuring adequate performance in replacing a TDM link with IP 
is to assure that adequate idle time is reserved for voice on the IP 
segment(s) involved in the route.  In this way, latency can be 
stabilized, and if maintained below a certain (arbitrary) threshold, 
performance can be deemed acceptable.

The first step, of course, is to assure that the virtual TDM allocation 
does not exceed the available IP bandwidth (so leave a margin, which is 
huge in the example given).  The next step is to use routers which 
respect the TOS field (however it is used; diffserv/whatever), and 
finally, to assure that no non-VoIP traffic can be injected into the 
path with higher routing priority.

On a point-to-point link, a pair of typical Linux boxes can do all 
this.  Given the original problem, I would place Asterisk boxes at 
either end of the link, and have them blend the ordinary traffic with 
the VoIP traffic (which would probably use IAX to relay calls between 
the T1s), while assuring (enforcing) that VoIP packets are marked as 
highest priority.  There are varied ways of accomplishing this, and a 
good reference which I've used in the past can be found at:

http://www.lartc.org/lartc.html

Additionally, I think one could use the tunneling  techniques described 
in that guide to encapsulate the non-VoIP traffic such that its packets' 
originally marked TOS values are preserved for transit outside the 
segment used for TDM emulation.  In this way, that part of the segment 
bandwidth not required for VoIP would function as a dedicated link, 
allowing other prioritization of traffic such as interactive vs. bulk 
(or even other voice!), with the added advantage that it could use the 
reserved VoIP bandwidth when it is otherwise not required (albeit in the 
case of a T-1 over 10Mb, that's insignificant).

Is this easier or harder than TDMoE as described?  Does the TDMoE shared 
idle bandwidth?  What about stability (I'm thinking of SW releases)?  
What other drawbacks or advantages are there?

Date: Wed, 25 Jan 2006 23:53:59 -0700
From: Damon Estep [EMAIL PROTECTED]
Subject: [Asterisk-Users] * point to point t1 solution?
To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com

Can anyone point me to a reference or sample config for bypassing a
nailed up (point to point) t1 between two PBXs with asterisk and a pair
of t1 cards?
 
Right now I have 2 Nortel norstars connected to each other via a leased
line t1. I also have a solid 10mbps low latency microwave link between
the 2 sites.
 
My goal is to run an asterisk box at each end with a t1 card and
Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in a
relatively dumb configuration, with the goal of migrating off of the
norstars eventually.

 In past situations I would have done this with a pair of Cisco routers
with T1 interfaces in them but in this case I want to get asterisk into
the picture as an eventual replacement for the norstars.

 
  


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[Asterisk-Users] Fail over to Pri on VoIP connection failure

2006-01-26 Thread Cavanna, Richard

I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up).  Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens.  Either by time
limit by a failure event?

Any point in the right direction would be great

Thanks,


CLI output (cleansed to protect the innocent)

-- Executing Dial(Zap/47-1,
IAX2/VoIPServicePrividerOUT/011) in new stack
-- Called VoIPServicePrividerOUT/011
-- Call accepted by 72.34.43.5 (format g729)
-- Format for call is g729
-- Channel 0/23, span 2 got hangup request here I get a busy
signal
-- Hungup 'IAX2/ VoIPServicePrividerOUT-1'



[Outbound context]
exten = _9011.,1,Macro(dialout-trunk,4,${EXTEN:1},) 
exten = _9011.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9011.,3,Macro(outisbusy)  ; No available circuits
exten = _918.,1,Macro(dialout-trunk,2,${EXTEN:1},); 800 numbers to the
PRI
exten = _918.,2,Macro(outisbusy)   ; No available circuits
exten = _9Z.,1,Macro(dialout-trunk,4,${EXTEN:1},)
exten = _9Z.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9Z.,3,Macro(outisbusy); No available circuits


Richard 
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Re: [Asterisk-Users] * point to point t1 solution?

2006-01-26 Thread Matt Riddell (IT)

Damon Estep wrote:
 saw those, according to RAD they occupy 2mbps even when idle. about $750/each 
 for t1

Are you basically looking to make a T1 repeater?

Or is there simply something that is removed from the signalling by
Asterisk that you want to maintain?

-- 
Cheers,

Matt Riddell
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[Asterisk-Users] RE: RE: RE: IAX Provider

2006-01-26 Thread Kaleb L. Kunzler

This is Kaleb, I have ABSOLUTELY no ties whatsoever with any VOIP service or
product, I am just an end-user.  I currently use sixTel, I love it.  I have
tried others and had very bad experiences, sellvoip.net being the absolute
worst from my experience.   I haven't ever tried any of the unlimited
services like broadvoice, I only have tried the pay-as-you-go providers that
support the IAX protocol.  sixTel has been really good to me, they are a tad
slow on email support, but of you catch the through MSN messenger
([EMAIL PROTECTED]) they usually are pretty good.  I haven't dealt with their
service department for about over a month, I have had an account with them
for 2 months and had only one problem with inbound calls, their DIDs stopped
working for a small spell due to a problem with their carrier, beyond their
control (at least that is what they told me).  When configuring their
service it was 100% required that in iax.conf the context was sixTel with
the capital T or inbound wouldn't work, can't say that I have ever noticed a
problem with outbound.  I will send a copy of my sterilized sixTel config to
anyone that would like.  

No Dan, I do not have a tie with them, I am just a happy customer.  I
especially like the fact that they email me automatically if they aren't
able to reach my server when someone calls. (has happened a time or two, was
MY fault).  They also allow you to set up a fail-back number (can be pstn or
cellular or whatever) if your server is unreachable by them.  They win my
business;  If you don't like them, that is your call. 

Kaleb 

 

 I guess Kaleb has a tie up with them.. 

 

I could never get my sixtel try call out except once. They customer service
sucks as usual

 

I regret paying to them...Their sample config are all done from my side my
my calls comes out with 'NO ANSWER

 

Dan

 

On 25/01/06, Dovid Bender [EMAIL PROTECTED] wrote: 

Let me guess you have no affiliation with them what so
ever ? no commision on accounts either ?
--- Kaleb L. Kunzler 
[EMAIL PROTECTED] wrote:

 I use iax.cc and find their service to be superior
 to ANY other VOIP
 provider I have tried.  Their prices are 
 competitive, My calls always go
 through, I always get my calls, I couldn't think of
 a better provider.  They
 are picky with the context that you use in your
 IAX.cc, but as long as you 
 use the sample config that they provide it works
 beautifully. You only need
 $5 to open the account, that really isn't bad at
 all, others like
 sellvoip.net  http://sellvoip.net (bad) require $25 to open an account.
 If you need help getting
 their service to work for you, please contact me off
 list.


 On 1/25/06, [EMAIL PROTECTED] [EMAIL PROTECTED]
 wrote:
 
  You can try with www,iax.cc too but i guesst not
 luck with a test
  account.. 
 
  Dan
 
 
  On 25/01/06, Nilesh Londhe [EMAIL PROTECTED]
 wrote:
  
   I use www.voipjet.com and find it OK.
  
   On 1/24/06, Roberto Pereyra 
 [EMAIL PROTECTED] wrote: 
Hi

   
I looking a good IAX service for a emerging
 voip provider.
   
Better with a test account to try. 
   
Thanks in advance.
   
roberto

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Message: 10
Date: Thu, 26 Jan 2006 14:55:22 +
From: bails [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TDM400 pinout
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii; format=flowed

BJ Weschke wrote:
 On 1/26/06, bails [EMAIL PROTECTED] wrote:
 
Chris Bagnall wrote:

Hi I'm looking for a pinout for the above.  Note this has
what i'd call
RJ45 sockets (or someone smart can correct me).  I need to
plug into BT (rj13?).


Are you sure the TDM400 has RJ45 sockets? The pair I've got here have
RJ12
sockets.

I assume with the mention of BT, you're in the UK. The line is on pins
2+5
of the BT connector, which'd usually translate to the 2 inner pins of an
RJ11 connector (pins 2+3). You should find an old modem cable will do the
job fine.

If your TDM400 really does have RJ45 sockets, then you'd expect the line
to
be on the middle pins (pins 4+5), similar to a modtap used in structured

Re: [Asterisk-Users] Cannot compile chan_bluetooth on Asterisk 1.2.1

2006-01-26 Thread Nilesh Londhe

Thanks a billion.

Outbound bluetooth dialling on the lines of
Dial(BLT/DevName/8005551212) worked for me.

Still trying out the inbound route. Before I created the [bluetooth]
context, it tried to reach the [default] context but then I began by
creating a new context [bluetooth] in extensions.conf and got my
internal SIP phone to ring when I received a call on my SE T616 cell
phone. However, I could not get the inbound line answered and I will
continue to work on this over the weekend and report back my progress.

On 1/25/06, Joseph Tanner [EMAIL PROTECTED] wrote:
 Again, my documentation is still sparse.  I should have noted that the
 phone will recognize asterisk and connect even if the channel in
 bluetooth.conf is configured wrong.  You'll just get no audio, or
 disconnects, or what-not until it's set correctly.  So realize that
 later on when you're testing.  Also the usb dongle must have a CSR
 chipset, else it won't work (well, at least probably won't work, I'll
 provide instructions on how to tell if it should work or not later).

 Here's the relevant instructions on
 http://www.crazygreek.co.uk/content/chan_bluetooth for how to dial
 out:  Send a call out by using Dial(BLT/DevName/0123456).

 As far as dialing in, there's a special context (I think [bluetooth]
 maybe?  I'll have to get back to you on that).  I know that it should
 work fine, because I tried dialing the phone, asterisk picked it up
 then immediately disconnected because there was no context for it to
 go to (I think it tried to fall back on [default], which I didn't have
 configured to accept an incoming call).

 Good luck!

 Joseph Tanner

 On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote:
  Thanks a lot. I succeeded in pairing my Sony Ericson T616 using your
  instructions at
  http://www.thetechguide.com/howto/asterisk/chanbluetooth.html without
  any problems. I rebooted and the phone prompted me to connect to
  asterisk. I provided the pin 1234 and voila it connected...
 
  Couple of observations:
 
  I started off with clean slate and booted off from [EMAIL PROTECTED] 2.2 CD.
  skipped the initial yum -u update part to save some time.
 
  When I ran the sdptool search --bdaddr MACADDRESS 0x111F command,
  below is what I got:
 
  Class 0x111F
  Searching on MACADDRESS
  Service Name: HF Voice Gateway
  Service RecHandle: 0x10007
  Service Class ID List:
   (0x111f)
  Generic Audio (0x1203)
  Protocol Descriptor List:
  L2CAP (0x0100)
  RFCOMM (0x0003)
  Channel: 6
  Profile Descriptor List
   0x111e
  Version 0x0100
 
  Note that in /etc/asterisk/bluetooth.conf, I kept Channel = 3 (did not
  change it to 6) and it paired my tooth in the first attempt after I
  rebooted asterisk box.
 
  Now, I want to get rid of my Doc-N-Talk that I currently connect my
  T616 to and the other end of Doc-N-Talk goes to x100p.
 
  Although I have worked with linux a bit, I am basically an ASTERISK
  NEWBIE so please pardon my ignorane but I don't know what to do
  next...that is.. how to define this bluetooth channel to make and
  receive calls using this setup...
 
  Appreciate your help.
 
 
  On 1/25/06, Joseph Tanner [EMAIL PROTECTED] wrote:
   Please note this is a work in progress:
  
   http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
  
   Basically the bluetoothfiles.tar.gz has the cvs code with the Makefile
   that worked for me, plus the edited Makefile in
   /usr/src/asterisk/channels, and the bluez edits I needed (hcid.conf
   with the correct profile, the files needed for the pin which is set to
   1234, etc.).  The guide is supposed to walk a person through the
   entire process of getting an Asterisk box setup and bluetooth working,
   but it's grossly incomplete.  Maybe it'll help you out.
  
   Joseph Tanner
  
   On 1/25/06, Nilesh Londhe [EMAIL PROTECTED] wrote:
Hi Joseph:
   
I still couldn't compile the newest cvs version of chan_bluetooth, so
I again tried my trick of using the Makefile from an older version
(which used .tmp to compile) and it worked!
   
Can you please point to the cvs you used, the location and content of
pin files you created and paste a copy of the make file that worked
for you?
   
Appreciate you sharing this information. Thanks.
   
On 1/20/06, Joseph Tanner [EMAIL PROTECTED] wrote:
 Ok, I did get this going (somewhat), and in case someone else has the
 same issues I'll detail what I had to do.

 First, I was using the instructions at
 http://mundy.org/blog/index.php?p=79.  They stated that [EMAIL 
 PROTECTED]
 2.2 already had all the rpms necessary for bluetooth and that I could
 skip the yum install step.  I did, however, run the command anyways
 after a few failed attempts.  There's an error in the rpm name, they
 tell you to install bluez-libs, the correct name is bluez-libs-devel
 (at least, that's what I needed to install).

 I still couldn't compile the newest cvs version of

Re: [Asterisk-Users] Cannot compile chan_bluetooth on Asterisk 1.2.1

2006-01-26 Thread Nilesh Londhe

BTW, I did get clear bidirectional audio when I succeded in dialing
out...(with the channel = 3 in /etc/asterisk/bluetooth.conf) I have
Sony Ericsson T616 connected to a cheap commodity bluetooth USB dongle
that I bought ages ago from meritline.

On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote:
 Thanks a billion.

 Outbound bluetooth dialling on the lines of
 Dial(BLT/DevName/8005551212) worked for me.

 Still trying out the inbound route. Before I created the [bluetooth]
 context, it tried to reach the [default] context but then I began by
 creating a new context [bluetooth] in extensions.conf and got my
 internal SIP phone to ring when I received a call on my SE T616 cell
 phone. However, I could not get the inbound line answered and I will
 continue to work on this over the weekend and report back my progress.

 On 1/25/06, Joseph Tanner [EMAIL PROTECTED] wrote:
  Again, my documentation is still sparse.  I should have noted that the
  phone will recognize asterisk and connect even if the channel in
  bluetooth.conf is configured wrong.  You'll just get no audio, or
  disconnects, or what-not until it's set correctly.  So realize that
  later on when you're testing.  Also the usb dongle must have a CSR
  chipset, else it won't work (well, at least probably won't work, I'll
  provide instructions on how to tell if it should work or not later).
 
  Here's the relevant instructions on
  http://www.crazygreek.co.uk/content/chan_bluetooth for how to dial
  out:  Send a call out by using Dial(BLT/DevName/0123456).
 
  As far as dialing in, there's a special context (I think [bluetooth]
  maybe?  I'll have to get back to you on that).  I know that it should
  work fine, because I tried dialing the phone, asterisk picked it up
  then immediately disconnected because there was no context for it to
  go to (I think it tried to fall back on [default], which I didn't have
  configured to accept an incoming call).
 
  Good luck!
 
  Joseph Tanner
 
  On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote:
   Thanks a lot. I succeeded in pairing my Sony Ericson T616 using your
   instructions at
   http://www.thetechguide.com/howto/asterisk/chanbluetooth.html without
   any problems. I rebooted and the phone prompted me to connect to
   asterisk. I provided the pin 1234 and voila it connected...
  
   Couple of observations:
  
   I started off with clean slate and booted off from [EMAIL PROTECTED] 2.2 
   CD.
   skipped the initial yum -u update part to save some time.
  
   When I ran the sdptool search --bdaddr MACADDRESS 0x111F command,
   below is what I got:
  
   Class 0x111F
   Searching on MACADDRESS
   Service Name: HF Voice Gateway
   Service RecHandle: 0x10007
   Service Class ID List:
(0x111f)
   Generic Audio (0x1203)
   Protocol Descriptor List:
   L2CAP (0x0100)
   RFCOMM (0x0003)
   Channel: 6
   Profile Descriptor List
0x111e
   Version 0x0100
  
   Note that in /etc/asterisk/bluetooth.conf, I kept Channel = 3 (did not
   change it to 6) and it paired my tooth in the first attempt after I
   rebooted asterisk box.
  
   Now, I want to get rid of my Doc-N-Talk that I currently connect my
   T616 to and the other end of Doc-N-Talk goes to x100p.
  
   Although I have worked with linux a bit, I am basically an ASTERISK
   NEWBIE so please pardon my ignorane but I don't know what to do
   next...that is.. how to define this bluetooth channel to make and
   receive calls using this setup...
  
   Appreciate your help.
  
  
   On 1/25/06, Joseph Tanner [EMAIL PROTECTED] wrote:
Please note this is a work in progress:
   
http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
   
Basically the bluetoothfiles.tar.gz has the cvs code with the Makefile
that worked for me, plus the edited Makefile in
/usr/src/asterisk/channels, and the bluez edits I needed (hcid.conf
with the correct profile, the files needed for the pin which is set to
1234, etc.).  The guide is supposed to walk a person through the
entire process of getting an Asterisk box setup and bluetooth working,
but it's grossly incomplete.  Maybe it'll help you out.
   
Joseph Tanner
   
On 1/25/06, Nilesh Londhe [EMAIL PROTECTED] wrote:
 Hi Joseph:

 I still couldn't compile the newest cvs version of chan_bluetooth, 
 so
 I again tried my trick of using the Makefile from an older version
 (which used .tmp to compile) and it worked!

 Can you please point to the cvs you used, the location and content of
 pin files you created and paste a copy of the make file that worked
 for you?

 Appreciate you sharing this information. Thanks.

 On 1/20/06, Joseph Tanner [EMAIL PROTECTED] wrote:
  Ok, I did get this going (somewhat), and in case someone else has 
  the
  same issues I'll detail what I had to do.
 
  First, I was using the instructions at
  http://mundy.org/blog/index.php?p=79.  They stated that [EMAIL

Re: [Asterisk-Users] RE: RE: RE: IAX Provider

2006-01-26 Thread [EMAIL PROTECTED]

KalebIm atleast happy to hear that you get what you have paid for. I had not been able to get tru with international calls ever since the service was taken. I ad informed them and it takes ages to reply. They are asking be weird questions and any response again will be after a century (so to say).
I'm using [EMAIL PROTECTED] Let me have the configs anyways... U can understand why i said that because im a victim. No hard feelings. You can see that its not only me who says bad about them.. and u r the only one who cheers them.
So lets try the best... I'm a lot happier with VOIPJET VOXEE...DanOn 26/01/06, Kaleb L. Kunzler

[EMAIL PROTECTED] wrote:This is Kaleb, I have ABSOLUTELY no ties whatsoever with any VOIP service or
product, I am just an end-user.I currently use sixTel, I love it.I havetried others and had very bad experiences,
sellvoip.net being the absoluteworst from my experience. I haven't ever tried any of the unlimited
services like broadvoice, I only have tried the pay-as-you-go providers thatsupport the IAX protocol.sixTel has been really good to me, they are a tadslow on email support, but of you catch the through MSN messenger
([EMAIL PROTECTED]) they usually are pretty good.I haven't dealt with theirservice department for about over a month, I have had an account with them
for 2 months and had only one problem with inbound calls, their DIDs stopped
working for a small spell due to a problem with their carrier, beyond theircontrol (at least that is what they told me).When configuring theirservice it was 100% required that in iax.conf the context was sixTel with
the capital T or inbound wouldn't work, can't say that I have ever noticed aproblem with outbound.I will send a copy of my sterilized sixTel config toanyone that would like.No Dan, I do not have a tie with them, I am just a happy customer.I
especially like the fact that they email me automatically if they aren'table to reach my server when someone calls. (has happened a time or two, wasMY fault).They also allow you to set up a fail-back number (can be pstn or
cellular or whatever) if your server is unreachable by them.They win mybusiness;If you don't like them, that is your call.Kaleb I guess Kaleb has a tie up with them..
I could never get my sixtel try call out except once. They customer servicesucks as usualI regret paying to them...Their sample config are all done from my side my
my calls comes out with 'NO ANSWERDanOn 25/01/06, Dovid Bender
[EMAIL PROTECTED] wrote:
Let me guess you have no affiliation with them what soever ? no commision on accounts either ?--- Kaleb L. Kunzler
[EMAIL PROTECTED]
wrote: I use iax.cc and find their service to be superior to ANY other VOIP provider I have tried.Their prices are competitive, My calls always go through, I always get my calls, I couldn't think of
a better provider.They are picky with the context that you use in your IAX.cc, but as long as you use the sample config that they provide it works beautifully. You only need
$5 to open the account, that really isn't bad at all, others like sellvoip.net
http://sellvoip.net (bad) require $25 to open an account.
If you need help getting their service to work for you, please contact me off list. On 1/25/06,
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote: You can try with www,iax.cc too but i guesst not luck with a test account.. Dan

On 25/01/06, Nilesh Londhe [EMAIL PROTECTED] wrote: I use
www.voipjet.com and find it OK.
On 1/24/06, Roberto Pereyra [EMAIL PROTECTED]
wrote:Hi
I looking a good IAX service for a emerging voip provider. Better with a test account to try. Thanks in advance.
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[Asterisk-Users] [Fwd: Asterisk as an Ascend box]

2006-01-26 Thread Etienne Pretorius


Sorry not sure the mail was sent to the correct address:

--
Kind Regards
Etienne

---BeginMessage---

Hello all,

I was just wandering if it is possible to make Asterisk become a 
replacement for an Ascend box and then utilise the unused channels to 
make outgoing and/or incoming calls?
Possibly use TDMoE for multiple replacement boxes. Can anyone give me 
some insight on this.


--
Kind Regards
Etienne



---End Message---
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RE: [Asterisk-Users] * point to point t1 solution? / alternatives

She ain't cheap, but this'll work:

http://www.blackboxcanada.com/Catalog/Detail.aspx?cid=381mid=4291

It's TDMoIP so 2 T1 boxes tied together should work like this:

T1--TDMXX card--Asterisk--TDMXX card--Voice Mux--Broadband--Voice Mux--TDMXX
card --Asterisk

at about $7K Cdn it'd be worthwhile to rewrite a dialplan to use IAX
instead. 
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RE: [Asterisk-Users] Snom360 Sidecar Asterisk




Snom360 with Sidecar works perfectly. THe Cisco expnsion I 
have yet to make work. I'll sell it to you if you want ( :-) 
)


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of c 
  waddySent: Thursday, January 26, 2006 10:31 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] Snom360 Sidecar  Asterisk
  
  We are looking to replace our existing Legacy PBX with Asterisk. Our 
  receptionist currently has a light display for a certain extension when 
  someone is on a call. When she needs to transfer she simply hits that button. 
  
  
  Is it possible to use a snom360 + Sidecar to monitor 30 extensions and 
  make transfers using the buttons? Does the Cisco 7960 expansion module work 
  with asterisk?
  
  Thanks.
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[Asterisk-Users] Plea to support a much needed function for Call Centers in Asterisk.

I have contacted Digium and have received a quote of $7,000US to
implement what I will refer to as 'whisper mode'.

It will allow a person to speak to only one side of a bridged call. For
example, I am using ChanSpy to listen to an agent and what they are
hearing and saying. But I cannot tell the agent something, without
calling them on another line. This wil allow you to speak to your agent
without the customer hearing.

I have already put up 1,000US for this. I just need 6,000 more. 

I know many have asked for this before, now is our chance to do this. 

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[Asterisk-Users] CDR logging in /var/log/asterisk instead of MySQL DB

2006-01-26 Thread Michaël Gaudette




Hi,

I've just 
reinstalled Asterisk 1.2.3 on a fresh system and since I've noticed that the CDR 
logging in MySQL (on a different computer) has stopped. I thought it 
wasn't logging anything at all, but I realized after a bit of searching that 
there were log files in /var/log/asterisk/cdr_customand 
/var/log/asterisk/cdr_csv with up to date logs.

My cdr_mysql.conf is 
set up properly, and I get no indication that the connection to MySQL is not 
working properly. Has something changed since 1.2.2 or 
1.2.3?

Regards,

Mike


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[Asterisk-Users] Dynamically disabling echo cancellation (Zap).

2006-01-26 Thread Ken D'Ambrosio

Hi!  For reasons that I won't bore people with, I'd like to disable echo
cancellation on-the-fly, depending on which DID a call came in on.  I've
seen things like spandsp disable EC for faxes, so I know it's possible. 
Any idea where to start looking?  (I assume I'll have to make a helper
application of some sort to be called externally, and that's fine.)

Thanks!

-Ken

P.S.  If this question is more appropriate for a developer's list, please
let me know.

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Re: [Asterisk-Users] * point to point t1 solution?

2006-01-26 Thread Don Pobanz


Damon Estep wrote:

Thanks Matt,
 
PRI signalling means that calls and answered and dialed (aka signalled) by asterisk, the goal is to maintain the signalling between the two nortel boxes.
 
I have gathered that raw point to point circuit emulation is not possible on asterisk...
 


To connect the channels of the T1 straight through would be by using a 
Digital Access Cross Connect system (DACS in proprietary ATT lingo). I 
believe there is this capability in zaptel, though this does not seem 
like the best option. As mentioned before, the ISDN PRI signaling seems 
like a much better solution.



I am aware of how to connect a PBX to asterisk using ISDN PRI signalling.


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[Asterisk-Users] Local Channel Call Looping

2006-01-26 Thread Darren Sessions

*** If anyone has a better way of doing this, please post to the list. I 
hadn't seen anything on this list or in channel.c/chan_local.c - which 
prompted this email ***


I'm not sure how many VoIP providers out there are using Asterisk as a 
service platform like we do, but I thought I'd share an experience with 
call looping that was racking my brain with the list.


One of the features we offer our customers, is of course, call 
forwarding. We take a call in and spit it back out to whatever the call 
forward number is set by the customer.


With our particular proxy setup, if a call originates from * to the 
proxy it will never loop back to *; this prevents SIP call loops.


In *, for an on-net call forward number, we would use the dial command 
to call (if it was registered) the customer's device with SIP via the 
proxy, and also dial a local channel to process any of the 'forward to' 
customer's features; again, this is for an on-net call.


The problem was that if when we dial the local channel and that customer 
had forwarded calls to the first number or calls were setup to forward 
from cust1 to cust2 to cust3 to cust1, we were getting an infinite local 
channel loop. As you can imagine, the load on * was off the charts.


The solution to the problem finally ended up being to set inherited 
channel variables.


First, we'd read/parse the channel variable to determine if the call was 
coming in anything other than a local channel. If it was, a variable 
with that called number label was immediately set to a value of 1 - i.e. 
the first in the chain.


Next, an addition variable with the 'call forward to' number was also 
given a value of 1, and then the call was processed. When the new local 
channel for the 'forward to' number was spawned, and assuming that call 
forwarding was set on that number, the process would repeat with this 
inherited variable label scheme.


The catch is that in each iteration at the same time the call forward to 
number is being labeled, the system would check that variable for a 
value before it tried to assign one. If the variable had a value, it was 
safe to assume that it had already been processed in the call chain 
somewhere and therefore the system would be looping the call if it 
continued.


Here are some sanitized Perl based AGI excerpts that accomplish this:


sub callfwd_loop_check
{
  my %v;
  ($v{callednum},$v{cfnum}) = @_;
  $v{num} = $AGI-get_variable($v{cfnum});
  if ($v{num})
  {
  debug( Call Loop Anaylsis for .$v{callednum}. = 
LOOPING);

  return(1);
  }
  else
  {
  debug( Call Loop Anaylsis for .$v{callednum}. = NO 
LOOP);

  $AGI-exec('Set',__.$v{cfnum}.=.$v{callednum})
  }
  return;
}

$AGI-exec('Set',__.$callednumber.=1) if ($calltype !~/^Local/);

if (callfwd_loop_check($callednumber,$callfwdtonum))
{
  return;
}

$AGI-exec('Dial',Local/+.$callfwdtonum.[EMAIL PROTECTED]SIP/+.$callfwdtonum.[EMAIL PROTECTED]|180); 




I hope this all makes sense! :)

Thanks,

- Darren
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RE: [Asterisk-Users] VOIP Router

2006-01-26 Thread Robert Augustyn

Arek,
Where can you get these?
robert 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Arek Bekiersz
 Sent: Thursday, January 26, 2006 7:50 AM
 To: [EMAIL PROTECTED]
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] VOIP Router
 
 Hi,
 
 
 Try one of Venus 2804, 2808 or 2832 from Tainet corporation.
 They support SIP or MGCP and they come with VPN.
 
 http://www.tainet.net
 Proceed to Product/VoIP/Venus
 
 --
 Regards,
 Arek Bekiersz
 
 
 
 Mohamed Farid wrote:
  Dear All :
  I need to link my HQ to some Remote Sites - I need a Router which 
  supports VOIP , and VPN Also the Router Should has 3 FXS 
 ports and 1 
  FXO ...
  The call should be routed from the Remote Site to the HQ 
 through a VPN 
  tunnel ( 3DES ) ...
  Any Advise ?
  Mohamed Farid ,,
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Re: [Asterisk-Users] Snom360 Sidecar Asterisk

2006-01-26 Thread Patrick

On Thu, 2006-01-26 at 15:31 +, c waddy wrote:
 We are looking to replace our existing Legacy PBX with Asterisk. Our
 receptionist currently has a light display for a certain extension
 when someone is on a call. When she needs to transfer she simply hits
 that button.
  
 Is it possible to use a snom360 + Sidecar to monitor 30 extensions and
 make transfers using the buttons? Does the Cisco 7960 expansion module
 work with asterisk?

About the Cisco 7960 expansion module, maybe have a look at
http://chan-sccp.berlios.de and their mailinglist archives.

Regards,
Patrick

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Re: [Asterisk-Users] Re: Random Disconnects

2006-01-26 Thread Thczv F. Thczv

On 1/26/06, Tomislav Parcina [EMAIL PROTECTED] wrote:

Hi Tomislav,

  I am not very satisfied with this, though.  I want to use some
  features (like Park) that apparently don't work well with reinvites.
  Have any of the rest of you had any luck troubleshooting this problem?

 Your RTP stream doesn't pass thrue Asterisk and it can't hear that you
 have pressed any key (that you are requesting that he parks the call).

I understand that.  I have a different problem:  My calls randomly get
disconnected when asterisk is in the media path.  So, for now I have
tried to take asterisk out of the media path.  Not being able to park
is a consequences of that.  What I really want to do is figure out why
my calls get disconnected.  If I could fix that, I could disable
reinvites and use park again.

Dave
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RE: [Asterisk-Users] Dynamically disabling echo cancellation (Zap).

 You can do a down and dirty test to see if it will work.  You can
record the start of a fax tone into a file.

Then after you answer the channel play the file. The 'special tone' will
cancel all of the Ecs on the line.

Its dity but will work in a pinch.


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ken D'Ambrosio
 Sent: Thursday, January 26, 2006 12:00 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Dynamically disabling echo 
 cancellation (Zap).
 
 Hi!  For reasons that I won't bore people with, I'd like to 
 disable echo cancellation on-the-fly, depending on which DID 
 a call came in on.  I've seen things like spandsp disable EC 
 for faxes, so I know it's possible. 
 Any idea where to start looking?  (I assume I'll have to make 
 a helper application of some sort to be called externally, 
 and that's fine.)
 
 Thanks!
 
 -Ken
 
 P.S.  If this question is more appropriate for a developer's 
 list, please let me know.
 
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[Asterisk-Users] addmailbox script

2006-01-26 Thread Tim Leeland









What happened to the addmailbox script in version 1.2.3?






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[Asterisk-Users] app_background and app_cepstral

2006-01-26 Thread Jason Wolfe

currently, when using swift TTS engine with app_cepstral, generated audio is
streamed to the channel.  This means that a call to ceptsral operates like
app_playback.  I need the functionality of app_background.  I'm thinking I
have two options... 1.) use system() to call swift engine, create a
temporary file, background() the temp file, and then delete the temp file.
option 2.) write this functionality into app_cepstral and use a flag in the
call.

Has anyone already solved this problem, or does anyone have another
suggestion?

Jason

Jason


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RE: [Asterisk-Users] addmailbox script




Don't 
need it. Add entries in voicemail.conf and mailbox is created on the 
fly...

  -Original Message-From: Tim Leeland 
  [mailto:[EMAIL PROTECTED]Sent: Thursday, January 26, 2006 10:47 
  AMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] addmailbox script
  
  What happened to the addmailbox 
  script in version 
1.2.3?
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RE: [Asterisk-Users] addmailbox script