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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>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.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> Douglas Garstang
[mailto:dgarstang@oneeighty.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Saturday, March 25, 2006
12:37 PM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] RE: IAX
Incoming/Outgoing</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<div>
<p class=MsoNormal><font size=2 color=blue face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:blue'>12 hours later... still playing with this.
Anyone got any ideas?</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<div>
<p class=MsoNormal><font size=2 color=blue face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:blue'>Doug.</span></font><o:p></o:p></p>
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<blockquote style='border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b> Douglas Garstang <br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, March 24, 2006 10:53
PM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial
Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> IAX Incoming/Outgoing</span></font><o:p></o:p></p>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>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.<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Has anyone got a working example they could supply? Can I do all this
with just three peers and one username?<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Thanks... Doug.<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>[pbx1_inbound]<br>
type=user<br>
auth=rsa<br>
inkeys=pbx1<br>
username=pbx1_inbound<br>
deny=0.0.0.0<br>
permit=xxx.187.142.203<br>
context=global_pbx_transfer<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>[pbx1_outbound]<br>
type=peer<br>
auth=rsa<br>
outkey=pbx1<br>
username=pbx1<br>
host=pbx1.ipt.yyy.com<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>[pbx2_inbound]<br>
type=user<br>
auth=rsa<br>
inkeys=pbx2<br>
username=pbx2_inbound<br>
deny=0.0.0.0<br>
permit=xxx.187.142.204<br>
context=global_pbx_transfer<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>[pbx2_outbound]<br>
type=peer<br>
auth=rsa<br>
outkey=pbx1<br>
username=pbx1<br>
host=pbx2.ipt.yyy.com<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>[pbx3_inbound]<br>
type=user<br>
auth=rsa<br>
inkeys=pbx3<br>
username=pbx3_inbound<br>
deny=0.0.0.0<br>
permit=xxx.187.142.234<br>
context=global_pbx_transfer<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>[pbx3_outbound]<br>
type=peer<br>
auth=rsa<br>
outkey=pbx1<br>
username=pbx3<br>
host=pbx3.ipt.yyy.com<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<blockquote style='margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=2
face="Times New Roman"><span style='font-size:10.0pt'>-----Original
Message----- <br>
<b><span style='font-weight:bold'>From:</span></b> George Vagenas
[mailto:gvagasterisk@gmail.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Fri 3/24/2006 10:30 PM <br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion <br>
<b><span style='font-weight:bold'>Cc:</span></b> George Vagenas <br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [Asterisk-Users] SIP
trunk problem</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Marty,<br>
<br>
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.<br>
<br>
<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 3/25/06, <b><span style='font-weight:bold'>Martin
Joseph</span></b> <<a href="mailto:ast@stillnewt.org">ast@stillnewt.org</a>>
wrote:</span></font></span> <o:p></o:p></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
On Mar 24, 2006, at 1:19 PM, George Vagenas wrote:<br>
<br>
> Hi all,<br>
><br>
> I have the following problem, working with a SIP provider, if i
setup<br>
> my SJPhone to register directly to their STUN server and working over <br>
> a 384/128 ADSL i have a really good quality, but then if i configure<br>
> Asterisk to register to the same provider over the same 384/128<br>
> circuit the quality is REALLY BAD. The obvious difference is that <br>
> using directly the SJPhone i am using STUN, while when i am using<br>
> Asterisk to connect to my SIP provider and the SJPhone to connect to<br>
> Asterisk i have the following configuration for Asterisk.<br>
><br>
><br>
> register => <a href="mailto:user:pass@sip.provider.com">user:pass@sip.provider.com</a><br>
><br>
> [mysip]<br>
> host=<a href="http://sip.provider.com">sip.provider.com</a><br>
> type=peer <br>
> qualify=yes<br>
> username=user<br>
> secret=pass<br>
> nat=yes<br>
> disallow=all<br>
> allow=ulaw<br>
><br>
><br>
> I am using Asterisk 1.2.3.<br>
><br>
> I think that i am missing something or misconfigure something
because <br>
> for sure its not matter of the ADSL since in both tests i am doing i<br>
> am using the same circuit.<br>
><br>
> Any idea please????<br>
I don't think using ulaw on a 128K bit upstream circuit is a good<br>
choice. I would use g729.<br>
<br>
Marty<br>
<br>
PS I can't be the stun server if asterisk is working, but quality is<br>
poor.<br>
<br>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
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</blockquote>
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