[Asterisk-Users] RE: IAX Incoming/Outgoing

Steve Totaro stotaro at asteriskhelpdesk.com
Sat Mar 25 13:57:35 MST 2006


No, you need to use different names.  You can use friend rather than
having separate entries for in/out.  What do you get when you type iax2
show peers?  You should be able to use friend and the same three entries
on each box with the exception of changing the IP addresses.

 

  _____  

From: Douglas Garstang [mailto:dgarstang at oneeighty.com] 
Sent: Saturday, March 25, 2006 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] RE: IAX Incoming/Outgoing

 

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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