[Asterisk-Users] RE: IAX Incoming/Outgoing

Steve Totaro stotaro at asteriskhelpdesk.com
Sat Mar 25 15:01:32 MST 2006


Just a few things Doug and they are just constructive criticism so don't
take them the wrong way. 

 

1.	You hijacked some else's thread about a SIP trunk problem.  Very
frowned upon and will decrease people willing to help..
2.	All of your posts are so dramatic and many times negative which
will also decrease willing help.
3.	You are posting way too much without experimenting and thinking
things through.  Take the list as a place to post knowledge and a place
to get answers when you have tried everything you can think of.

 

I had a rule to put your emails directly in my deleted items folder from
the first day you started posting to this list totally bashing asterisk
and the community.  I recently had to re-do my machine so the rule was
lost.  I am hoping that I don't have to put it back in place.

 

Now back to your problem.  

Simplify your conf.  Remove the keys and use secret=

Change your dial statement to Dial(iax2/username:secret at ipaddress)

Are you dialing from PBX3?

 

Thanks,

Steve Totaro

 

  _____  

From: Douglas Garstang [mailto:dgarstang at oneeighty.com] 
Sent: Saturday, March 25, 2006 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing

 

This is INSANE! My calling system has this iax.conf:

 

[pbx1]
type=friend
auth=rsa
inkeys=pbx1
outkey=pbx1
context=global_pbx_transfer
host=pbx1.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.203

 

[pbx2]
type=friend
auth=rsa
inkeys=pbx2
outkey=pbx1
context=global_pbx_transfer
host=pbx2.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.233

 

[pbx3]
type=friend
auth=rsa
inkeys=pbx3
outkey=pbx1
context=global_pbx_transfer
host=pbx3.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.234

 

and here's how I am dialling PBX2... as you can see I am dialling
_PBX2_:

exten => s-CHANUNAVAIL,1,Dial(IAX2/pbx2/${ARG1}@global_pbx_transfer,25,g
<mailto:IAX2/pbx2/$%7bARG1%7d at global_pbx_transfer,25,g> )

 

 

When I run an iax debug on the caller, I see

   VERSION         : 2
   CALLED NUMBER   : 2944099
   CODEC_PREFS     : (ulaw|g729)
   CALLING NUMBER  : 2944093
   CALLING PRESNTN : 0
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   CALLING NAME    : Foo

   LANGUAGE        : en
   CALLED CONTEXT  : global_pbx_transfer
   FORMAT          : 4
   CAPABILITY      : 65535
   ADSICPE         : 2
   DATE TIME       : 2006-03-25  11:24:58

 

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass:
ACK    
   Timestamp: 00005ms  SCall: 00004  DCall: 00006 [216.187.142.204:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass:
AUTHREQ
   Timestamp: 00010ms  SCall: 00004  DCall: 00006 [216.187.142.204:4569]
   AUTHMETHODS     : 4
   CHALLENGE       : 627190238
   USERNAME        : pbx3

 

What on gods green earth would possibly make asterisk want to send a
username of PBX3???

 

 

	-----Original Message-----
	From: Douglas Garstang 
	Sent: Saturday, March 25, 2006 11:16 AM
	To: Asterisk Users Mailing List - Non-Commercial Discussion
	Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing

	Well, right now I have this on box1:

	 

	[pbx1]
	type=friend
	auth=rsa
	inkeys=pbx1
	outkey=pbx1
	context=global_pbx_transfer
	host=pbx1.ipt.yyy.com
	deny=0.0.0.0
	permit=xxx.187.142.203

	 

	[pbx2]
	type=friend
	auth=rsa
	inkeys=pbx2
	outkey=pbx1
	context=global_pbx_transfer
	host=pbx2.ipt.yyy.com
	deny=0.0.0.0
	permit=xxx.187.142.233

	 

	[pbx3]
	type=friend
	auth=rsa
	inkeys=pbx3
	outkey=pbx1
	context=global_pbx_transfer
	host=pbx3.ipt.yyy.com
	deny=0.0.0.0
	permit=xxx.187.142.234

	 

	 

	 

	and this on box2:

	[pbx1]
	type=friend
	auth=rsa
	inkeys=pbx1
	outkey=pbx2
	context=global_pbx_transfer
	host=pbx1.ipt.yyy.com
	deny=0.0.0.0
	permit=xxx.187.142.203

	 

	[pbx2]
	type=friend
	auth=rsa
	inkeys=pbx2
	outkey=pbx2
	context=global_pbx_transfer
	host=pbx2.ipt.yyy.com
	deny=0.0.0.0
	permit=xxx.187.142.233

	 

	[pbx3]
	type=friend
	auth=rsa
	inkeys=pbx3
	outkey=pbx2
	context=global_pbx_transfer
	host=pbx3.ipt.yyy.com
	deny=0.0.0.0
	permit=xxx.187.142.234

	 

	And for some reason the calling system is sending pbx3 as the
username.... Why would it do that? 

	 

		-----Original Message-----
		From: Steve Totaro [mailto:stotaro at asteriskhelpdesk.com]
		Sent: Saturday, March 25, 2006 1:58 PM
		To: Asterisk Users Mailing List - Non-Commercial
Discussion
		Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing

		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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