[Asterisk-Users] RE: IAX Incoming/Outgoing

[Douglas Garstang]

12 hours later... still playing with this. Anyone got any ideas?
 
Doug.

-----Original Message-----
From: Douglas Garstang 
Sent: Friday, March 24, 2006 10:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: IAX Incoming/Outgoing


I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a call from pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say that pbx2 will look for a [pbx1_outbound] .... oh dear... this doesn't make sense any longer.
 
Has anyone got a working example they could supply? Can I do all this with just three peers and one username?
 
Thanks... Doug.
 
[pbx1_inbound]
type=user
auth=rsa
inkeys=pbx1
username=pbx1_inbound
deny=0.0.0.0
permit=xxx.187.142.203
context=global_pbx_transfer
 
[pbx1_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx1
host=pbx1.ipt.yyy.com
 
[pbx2_inbound]
type=user
auth=rsa
inkeys=pbx2
username=pbx2_inbound
deny=0.0.0.0
permit=xxx.187.142.204
context=global_pbx_transfer
 
[pbx2_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx1
host=pbx2.ipt.yyy.com
 
[pbx3_inbound]
type=user
auth=rsa
inkeys=pbx3
username=pbx3_inbound
deny=0.0.0.0
permit=xxx.187.142.234
context=global_pbx_transfer
 
[pbx3_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx3
host=pbx3.ipt.yyy.com
 

-----Original Message----- 
From: George Vagenas [mailto:gvagasterisk at gmail.com] 
Sent: Fri 3/24/2006 10:30 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: George Vagenas 
Subject: Re: [Asterisk-Users] SIP trunk problem


Marty,

But with the same 128 bit upstream circuit, directly connecting the SJPhone the Stun server and using ulaw,  everything is perfect. The problem comes when i am putting Asterisk in the picture.



On 3/25/06, Martin Joseph < ast at stillnewt.org> wrote: 


On Mar 24, 2006, at 1:19 PM, George Vagenas wrote:

> Hi all,
>
>  I have the following problem, working with a SIP provider, if i setup
> my SJPhone to register directly to their STUN server and working over 
> a 384/128 ADSL i have a really good quality, but then if i configure
> Asterisk to register to the same provider over the same 384/128
> circuit the quality is REALLY BAD. The obvious difference is that 
> using directly the SJPhone i am using STUN, while when i am using
> Asterisk to connect to my SIP provider and the SJPhone to connect to
> Asterisk i have the following configuration for Asterisk.
>
>
>  register => user:pass at sip.provider.com
>
>  [mysip]
>  host= sip.provider.com
>  type=peer 
>  qualify=yes
>  username=user
>  secret=pass
>  nat=yes
>  disallow=all
>  allow=ulaw
>
>
>  I am using Asterisk 1.2.3.
>
>  I think that i am missing something or misconfigure something because 
> for sure its not matter of the ADSL since in both tests i am doing i
> am using the same circuit.
>
>  Any idea please????
I don't think using ulaw on a 128K bit upstream circuit is a good
choice.  I would use g729.

Marty

PS I can't be the stun server if asterisk is working, but quality is
poor.

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