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<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=908293617-25032006>12
hours later... still playing with this. Anyone got any
ideas?</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=908293617-25032006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=908293617-25032006>Doug.</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B> Douglas Garstang
<BR><B>Sent:</B> Friday, March 24, 2006 10:53 PM<BR><B>To:</B> Asterisk Users
Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List -
Non-Commercial Discussion<BR><B>Subject:</B> IAX
Incoming/Outgoing<BR><BR></FONT></DIV>
<DIV>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.</DIV>
<DIV> </DIV>
<DIV>Has anyone got a working example they could supply? Can I do all this
with just three peers and one username?</DIV>
<DIV> </DIV>
<DIV>Thanks... Doug.</DIV>
<DIV> </DIV>
<DIV>[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</DIV>
<DIV> </DIV>
<DIV>[pbx1_outbound]<BR>type=peer<BR>auth=rsa<BR>outkey=pbx1<BR>username=pbx1<BR>host=pbx1.ipt.yyy.com</DIV>
<DIV> </DIV>
<DIV>[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</DIV>
<DIV> </DIV>
<DIV>[pbx2_outbound]<BR>type=peer<BR>auth=rsa<BR>outkey=pbx1<BR>username=pbx1<BR>host=pbx2.ipt.yyy.com</DIV>
<DIV> </DIV>
<DIV>[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</DIV>
<DIV> </DIV>
<DIV>[pbx3_outbound]<BR>type=peer<BR>auth=rsa<BR>outkey=pbx1<BR>username=pbx3<BR>host=pbx3.ipt.yyy.com</DIV>
<DIV> </DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV><FONT size=2>-----Original Message----- <BR><B>From:</B> George Vagenas
[mailto:gvagasterisk@gmail.com] <BR><B>Sent:</B> Fri 3/24/2006 10:30 PM
<BR><B>To:</B> Asterisk Users Mailing List - Non-Commercial Discussion
<BR><B>Cc:</B> George Vagenas <BR><B>Subject:</B> Re: [Asterisk-Users] SIP
trunk problem<BR><BR></FONT></DIV>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><BR>
<DIV><SPAN class=gmail_quote>On 3/25/06, <B class=gmail_sendername>Martin
Joseph</B> <<A href="mailto:ast@stillnewt.org">ast@stillnewt.org</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid"><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>_______________________________________________<BR>--Bandwidth
and Colocation provided by <A href="http://Easynews.com">Easynews.com</A>
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