Fw: Re: Re: [Asterisk-Users] inter Asterisk trunking IAX /IAX2

[Geo]

I originally wanted to answer with something ... tzarit and kevit
Readed probably before you invent rapid biz. I am asking to share any info 
/experience not your high spirit.
Thanks for less trivial answer,
G

>On Wed, Oct 05, 2005 at 12:44:27PM -0700, Thameem Ansari wrote:
>> I am using the inter asterisk trunking and the article in
>> voip-info.orgwill not work.
>
>I originally wanted to answer with something like: "is it on strike?" .
>If you want to get help here, please provide useful information to help
>with.
>
>Which article exactly? Could you please give the full URL?
>
>On what system(s) did you try it? What exactly did you do? What did
>happen? (error messages, CLI traces, etc.)
>
>How do you call from one Asterisk to another? What happens when you try
>to call? Could you provide the relevant config files?
>
>-- 
>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] inter Asterisk trunking IAX /IAX2

[Tzafrir Cohen]

On Wed, Oct 05, 2005 at 12:44:27PM -0700, Thameem Ansari wrote:
> I am using the inter asterisk trunking and the article in
> voip-info.orgwill not work.

I originally wanted to answer with something like: "is it on strike?" .
If you want to get help here, please provide useful information to help
with.

Which article exactly? Could you please give the full URL?

On what system(s) did you try it? What exactly did you do? What did
happen? (error messages, CLI traces, etc.)

How do you call from one Asterisk to another? What happens when you try
to call? Could you provide the relevant config files?

-- 
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] inter Asterisk trunking IAX /IAX2

[Thameem Ansari]

I am using the inter asterisk trunking and the article in voip-info.org will not work.

On 10/5/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Wed, Oct 05, 2005 at 03:51:38PM -0800, Geo wrote:> Hi,>> Anyone using inter Asterisk trunking IAX /IAX2 ?> Thanks,Not sure about IAX (1), but IAX2 is widely used. Before asking trivial
questions you probably should take the time reading about it inhttp://voip-info/wiki-Asterisk and similar places, though.--Tzafrir Cohen | 
[EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il
|  
| a Mutt's[EMAIL PROTECTED]
|  
|  bestICQ#
16849755
|  
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Re: [Asterisk-Users] inter Asterisk trunking IAX /IAX2

[Tzafrir Cohen]

On Wed, Oct 05, 2005 at 03:51:38PM -0800, Geo wrote:
> Hi,
> 
> Anyone using inter Asterisk trunking IAX /IAX2 ?
> Thanks,

Not sure about IAX (1), but IAX2 is widely used. Before asking trivial
questions you probably should take the time reading about it in
http://voip-info/wiki-Asterisk and similar places, though.

-- 
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] inter Asterisk trunking IAX /IAX2

[Kevin Walsh]

Geo [EMAIL PROTECTED] wrote:
> Anyone using inter Asterisk trunking IAX /IAX2 ?
>
No - you're the first to think of that.  Congratulations.

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Re: [Asterisk-Users] inter-asterisk meetme

[Steve Totaro]

I haven't played with it but I think you want to stream using ices.


- Original Message - 
From: "Zen Kato" <[EMAIL PROTECTED]>
To: ; <[EMAIL PROTECTED]>
Sent: Wednesday, August 03, 2005 6:20 PM
Subject: Re: [Asterisk-Users] inter-asterisk meetme


>
> "William Boehlke" wrote  :
>
> >
> > Why do you want to do that?
> >
> 100 sip users are connected to each CPU(P4 3.0MHz)s, then I would like to
> broadcast from one sip phone to 500 sip users. If I have 5 microphones
> in front of me, I can talk to 5 microphones, then 500 users can
> listen(one-way mode) simultaneously. But that is not elegant.
>
> --
> Zen
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Zen Kato
> > Sent: Wednesday, August 03, 2005 5:41 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] inter-asterisk meetme
> >
> > Hi,
> >
> > If there are 5 asterisk servers on the local net and each server runs
> > meetme, eg. 3311,3321,3331,3341,3351 respectively.
> >
> > Can I connect these 5 meetme conferences to one meetme using IAX2?
> >
> > Regards,
> >
> > Zen
> > ___
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> > Checked by AVG Anti-Virus.
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> >
> >
> > -- 
> > No virus found in this outgoing message.
> > Checked by AVG Anti-Virus.
> > Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005
> >
> >
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Re: [Asterisk-Users] inter-asterisk meetme

[Zen Kato]

"William Boehlke" wrote  :

> 
> Why do you want to do that? 
>  
100 sip users are connected to each CPU(P4 3.0MHz)s, then I would like to 
broadcast from one sip phone to 500 sip users. If I have 5 microphones
in front of me, I can talk to 5 microphones, then 500 users can
listen(one-way mode) simultaneously. But that is not elegant.

--
Zen

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Zen Kato
> Sent: Wednesday, August 03, 2005 5:41 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] inter-asterisk meetme
> 
> Hi,
> 
> If there are 5 asterisk servers on the local net and each server runs
> meetme, eg. 3311,3321,3331,3341,3351 respectively.
> 
> Can I connect these 5 meetme conferences to one meetme using IAX2?
> 
> Regards,
> 
> Zen
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> --
> No virus found in this incoming message.
> Checked by AVG Anti-Virus.
> Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005
>  
> 
> -- 
> No virus found in this outgoing message.
> Checked by AVG Anti-Virus.
> Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005
>  
> 
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RE: [Asterisk-Users] inter-asterisk meetme

[William Boehlke]

Why do you want to do that? 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zen Kato
Sent: Wednesday, August 03, 2005 5:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] inter-asterisk meetme

Hi,

If there are 5 asterisk servers on the local net and each server runs
meetme, eg. 3311,3321,3331,3341,3351 respectively.

Can I connect these 5 meetme conferences to one meetme using IAX2?

Regards,

Zen
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--
No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005
 

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005
 

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Re: [Asterisk-Users] inter asterisk

[Ousmane Doukara]

I think it has to do with my ZAP interface. Before my
DIAL(ZAP/1/51412345678) I have a Playback(message-transfert) which play
nicely. As soon as the ZAP
start ringing the PSTN phone, i have that helicopter sound.


- Original Message -
From: "David J Carter" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Sunday, February 06, 2005 7:18 AM
Subject: RE: [Asterisk-Users] inter asterisk


One thing I do on remote sites is set up a soft phone so I can call myself,
this proves out the link and quality before anything else. DIAX id good for
this as you can connect to multiple sites, also good to see if you have
problems before anyone else calls you to say there is a problem.

It also helps in cases like this, if your return quality is good then the
possible fault lies with the ZAP interface.

Process of elimination, works for me every time.

Regards

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ousmane Doukara
Sent: 06 February 2005 08:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] inter asterisk


Hi,
I am trying to forward calls to another * server with IAX
Here is What I want to Do
1- Call SERVER1, let say at 51412345678
2- SERVER1 should transfer the call to SERVER2 in a remote location
3- SERVER2 Receive the call and transfer it to the PSTN number.

I have one X100P  card on each machine. What is happening is that when the
remote party picks up the phone, all he can hear
is a weird sound.

CONFIGS:

 SERVER1:
  zaptel.conf
   -
   ~ [channels]
   ~ language=fr
   ~ context=montréal
   ~ signalling=fxs_ks
   ~ usercallerid=yes
   ~ callwaiting=yes
   ~ threewaycalling=yes
   ~ transfer=yes
   ~ cancellforward=yes
   ~ echocancel=yes
   ~ echocancelwhenbridged=yes
   ~ echotraining=yes
   ~ relaxdtmf=yes
   ~ busydetect=yes
   ~ busycount=4
   ~ callprogress=yes
   ~ group=1
   ~ channel=>1
   -- (same for SERVER2)

  IAX.conf
   
   ~ [general]
   ~ bindport=4569
   ~ delayreject=yes
   ~ language=fr
   ~ allow=all
   ~ jutterbuffer=no
   ~ register => username:[EMAIL PROTECTED]
   ~ tos=lowdelay
   ~ autokill=yes
   ~
   ~ [quebec]
   ~ type=friends
   ~ username = username
   ~ password=password
   ~ context=montréal
   ~ host=Dynamic
   ~ secret = password
   ~ disallow = all
   ~ allow=ulaw
   ~ allow=gsm

  extensions.conf
   --(Same for SERVER2 but no
registration)
   ~ [general]
   ~ static=yes
   ~ writeprotect=yes
   ~ autofallthrough=yes
   ~ [montréal]
   ~ exten=>s,1,Answer
   ~ exten=>s,2,Playback(message-transfer)
   ~
exten=>s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]
al) ; always the same number
   ~ exten=>s,4,Hangup



My remote server receive the call, answer the line and then
Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the
phone,
all she can hear is a weird sound.
What am I doing wrong ?

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RE: [Asterisk-Users] inter asterisk

[David J Carter]

One thing I do on remote sites is set up a soft phone so I can call myself,
this proves out the link and quality before anything else. DIAX id good for
this as you can connect to multiple sites, also good to see if you have
problems before anyone else calls you to say there is a problem.

It also helps in cases like this, if your return quality is good then the
possible fault lies with the ZAP interface.

Process of elimination, works for me every time.

Regards

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ousmane Doukara
Sent: 06 February 2005 08:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] inter asterisk


Hi,
I am trying to forward calls to another * server with IAX
Here is What I want to Do
1- Call SERVER1, let say at 51412345678
2- SERVER1 should transfer the call to SERVER2 in a remote location
3- SERVER2 Receive the call and transfer it to the PSTN number.

I have one X100P  card on each machine. What is happening is that when the
remote party picks up the phone, all he can hear
is a weird sound.

CONFIGS:

 SERVER1:
  zaptel.conf
   -
   ~ [channels]
   ~ language=fr
   ~ context=montréal
   ~ signalling=fxs_ks
   ~ usercallerid=yes
   ~ callwaiting=yes
   ~ threewaycalling=yes
   ~ transfer=yes
   ~ cancellforward=yes
   ~ echocancel=yes
   ~ echocancelwhenbridged=yes
   ~ echotraining=yes
   ~ relaxdtmf=yes
   ~ busydetect=yes
   ~ busycount=4
   ~ callprogress=yes
   ~ group=1
   ~ channel=>1
   -- (same for SERVER2)

  IAX.conf
   
   ~ [general]
   ~ bindport=4569
   ~ delayreject=yes
   ~ language=fr
   ~ allow=all
   ~ jutterbuffer=no
   ~ register => username:[EMAIL PROTECTED]
   ~ tos=lowdelay
   ~ autokill=yes
   ~
   ~ [quebec]
   ~ type=friends
   ~ username = username
   ~ password=password
   ~ context=montréal
   ~ host=Dynamic
   ~ secret = password
   ~ disallow = all
   ~ allow=ulaw
   ~ allow=gsm

  extensions.conf
   --(Same for SERVER2 but no
registration)
   ~ [general]
   ~ static=yes
   ~ writeprotect=yes
   ~ autofallthrough=yes
   ~ [montréal]
   ~ exten=>s,1,Answer
   ~ exten=>s,2,Playback(message-transfer)
   ~
exten=>s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]
al) ; always the same number
   ~ exten=>s,4,Hangup



My remote server receive the call, answer the line and then
Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the
phone,
all she can hear is a weird sound.
What am I doing wrong ?

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Re: [Asterisk-Users] inter asterisk

[Rich Adamson]

> I am trying to forward calls to another * server with IAX
> Here is What I want to Do
> 1- Call SERVER1, let say at 51412345678
> 2- SERVER1 should transfer the call to SERVER2 in a remote location
> 3- SERVER2 Receive the call and transfer it to the PSTN number.
>  
> I have one X100P  card on each machine. What is happening is that when the 
> remote party picks 
up the phone, all he can hear
> is a weird sound.
>  
> CONFIGS:
>  
>  SERVER1:
>   zaptel.conf
>-
>~ [channels]
>~ language=fr
>~ context=montréal
>~ signalling=fxs_ks
>~ usercallerid=yes
>~ callwaiting=yes
>~ threewaycalling=yes
>~ transfer=yes
>~ cancellforward=yes
>~ echocancel=yes
>~ echocancelwhenbridged=yes
>~ echotraining=yes
>~ relaxdtmf=yes
>~ busydetect=yes
>~ busycount=4
>~ callprogress=yes
>~ group=1
>~ channel=>1
>-- (same for SERVER2)
>  
>   IAX.conf
>
>~ [general]
>~ bindport=4569
>~ delayreject=yes
>~ language=fr
>~ allow=all
>~ jutterbuffer=no
>~ register => username:[EMAIL PROTECTED]
>~ tos=lowdelay
>~ autokill=yes
>~
>~ [quebec]
>~ type=friends
>~ username = username
>~ password=password
>~ context=montréal
>~ host=Dynamic
>~ secret = password
>~ disallow = all
>~ allow=ulaw
>~ allow=gsm
>  
>   extensions.conf
>--(Same for SERVER2 but no 
> registration)
>~ [general]
>~ static=yes
>~ writeprotect=yes
>~ autofallthrough=yes
>~ [montréal]
>~ exten=>s,1,Answer
>~ exten=>s,2,Playback(message-transfer)
>~ exten=>s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; 
> always 
the same number
>~ exten=>s,4,Hangup 
>  
>  
>  
> My remote server receive the call, answer the line and then 
> Dial(ZAP/1/51412345678). So far so 
good. But when 51412345678 pickup the phone,
> all she can hear is a weird sound.
> What am I doing wrong ?


Difficult to tell without some feedback from the CLI. If you actually
copy/pasted the above config statements, I'm assuming you manually
added all those "~" at the front of each line. If they are actually
in your config, get rid of them.

The statement "jutterbuffer=no" should be jitterbuffer=no.

One thing you might try to at least eliminate possible problems is
to change iax.conf to disallow=all and allow=gsm only. Get rid of
the allow=ulaw and do another test. Might as well add trunk=no 
to this link as well. (Must stop and restart * after making these 
type changes.)

You might try 'iax2 debug' from the CLI on both machines and look at
the detail to see if you can spot any conflicts or problems.



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Re: [Asterisk-Users] Inter-Asterisk Exchange

[M3 Freak]

On Fri, 2004-07-02 at 16:22, Bryan Brannigan wrote:
> > Depending on what you are planning to do in the datacenter you could
> > just put SIP phones/ATAs there rather than a full Asterisk server but
> > that would require some care in configuring your firewall.
> 
> Actually the users are will be remote to the datacenter.  The IPs in our
> office are dynamic so I imagine that would be an issue to just hosting the
> box there.

To prevent yourself from losing hair, one side, preferably the main
office, should have static IPs.  Then the remote users could connect to
the main office through a VPN, and viola, they're sitting on the same
network as the main office.  But, the remote users wouldn't need an
Asterisk server.  It wouldn't matter if their IPs changed because the
VPN could be configured at either end to take that into account.

Kanwar
Systems Aligned Inc.
www.systemsaligned.com

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Re: [Asterisk-Users] Inter-Asterisk Exchange

[Bryan Brannigan]

> Depending on what you are planning to do in the datacenter you could
> just put SIP phones/ATAs there rather than a full Asterisk server but
> that would require some care in configuring your firewall.

Actually the users are will be remote to the datacenter.  The IPs in our
office are dynamic so I imagine that would be an issue to just hosting the
box there.


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Re: [Asterisk-Users] Inter-Asterisk Exchange

[M3 Freak]

On Fri, 2004-07-02 at 15:49, Bryan Brannigan wrote:
> I would like to setup 2 Asterisk boxes.  One would be located in our
> office behind the firewall and hooked up to our analog lines.  The other
> would be located in a remote datacenter and used for our remote employees
> to connect to.  I would like to be able to accept calls on the Office
> Asterisk server and route them to the Datacenter Asterisk server.  Is this
> possible?

Yes.

Kanwar
Systems Aligned Inc.
www.systemsaligned.com

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Re: [Asterisk-Users] Inter-Asterisk Exchange

[George Pajari]

Bryan:

> I would like to setup 2 Asterisk boxes.  One would be located in our
> office behind the firewall and hooked up to our analog lines.  The other
> would be located in a remote datacenter and used for our remote employees
> to connect to.  I would like to be able to accept calls on the Office
> Asterisk server and route them to the Datacenter Asterisk server.  Is this
> possible?

Trivial. We're doing just that for a client over an RF link to get phone
service to a building which would otherwise have to wait months for land
lines.

Depending on what you are planning to do in the datacenter you could just
put SIP phones/ATAs there rather than a full Asterisk server but that would
require some care in configuring your firewall.

George Pajari
netVOICE communications
www.netvoice.ca
www.ip-centrex.ca

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