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<DIV>I did use IAX2 but sound quality wasn't that good which codec are you using
with IAX2 ?</DIV>
<DIV> </DIV>
<DIV><BR><FONT face=Arial size=2>*********** REPLY SEPARATOR
***********<BR><BR>On 12/6/2005 at 9:22 PM Alvaro Parres wrote:</FONT></DIV>
<BLOCKQUOTE
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid">
<DIV>Why using SIP instead of IAX2 ???</DIV>
<DIV> </DIV>
<DIV>Only a question becouse i always use IAX</DIV>
<DIV> </DIV>
<DIV><BR><BR> </DIV>
<DIV><SPAN class=gmail_quote>On 12/6/05, <B class=gmail_sendername>Waldo
Rubinstein</B> <<A
href="mailto:waldo@trianet.net">waldo@trianet..net</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Well...
not so perfectly.<BR><BR>What I'm experiencing is that during certain call
volumes, many calls<BR>go thru from box1 to box2. However, there are some
cases where I get<BR>this message:<BR><BR>Dec 6 11:11:19
WARNING[203]: chan_sip.c:9525 handle_response_invite:<BR>Forbidden - wrong
password on authentication for INVITE to <BR>'"5095551212" <<A
href="mailto:sip:5095551212@10.0.0.1">sip:5095551212@10.0.0.1</A>>;tag=as3e387d65'<BR><BR>and
the caller gets busy signal. However, other callers calling the<BR>same
number get thru with no problems. Why is this?
<BR><BR>Thanks,<BR>Waldo<BR><BR>On Dec 5, 2005, at 10:30 AM, Waldo
Rubinstein wrote:<BR><BR>> This worked perfectly.<BR>><BR>>
Thanks,<BR>> Waldo<BR>><BR>> On Dec 5, 2005, at 4:32 AM, xcel
wrote:<BR>> <BR>>><BR>>> Try this<BR>>><BR>>>
___________________________________<BR>>> 1st Machine
sip.conf<BR>>><BR>>> [box2]<BR>>>
username=box1<BR>>> type=friend<BR>>> host= <A
href="http://10.0.0.2">10.0.0.2</A><BR>>>
secret=*****<BR>>><BR>>> in
extensions.conf<BR>>><BR>>> exten =>
_XXXXXX,1,Dial(SIP/box2/${EXTEN})<BR>>><BR>>>
__________________________________ <BR>>> 2nd Machine
sip.conf<BR>>><BR>>> [box1]<BR>>>
username=box2<BR>>> type=friend<BR>>> host=<A
href="http://10.0.0.1">10.0.0.1</A><BR>>>
secret=*****<BR>>><BR>>> in extensions.conf<BR>>> exten
=> _XXXXX,1,Dial(SIP/box1/${EXTEN})<BR>>><BR>>>
--xce<BR>>><BR>>><BR>>> *********** REPLY
SEPARATOR ***********<BR>>><BR>>> On 12/5/2005 at
12:11 AM Waldo Rubinstein wrote: <BR>>><BR>>>> I have 2
Asterisk servers running 1.2.0. One of them is a PSTN<BR>>>>
gateway. Currently they are connected using IAX2. I wanted to
play<BR>>>> with SIP.<BR>>>><BR>>>> I setup a sip
entry (type=friend) in the PSTN gateway box and a sip <BR>>>> entry
(type=user) in the second box in order to send calls using
SIP<BR>>>> to the second box. This works fine. However, when I
setup the second<BR>>>> box as type=friend in order for it to be
able to send calls back to <BR>>>> the gateway box, then calls no
longer work from gateway box to the<BR>>>> second box. The reported
error is:<BR>>>><BR>>>> Dec 5 00:07:14
NOTICE[203]: chan_sip.c:9514 handle_response_invite: <BR>>>> Failed
to authenticate on INVITE to '"2125551212" <sip:<BR>>>> <A
href="mailto:2125551212@10.0.10.1">2125551212@10.0.10.1</A>>;tag=as0698b1b9'<BR>>>><BR>>>>
In the gateway box, my sip.conf looks like
this:<BR>>>><BR>>>> [general]<BR>>>>
allowguest=yes<BR>>>>
autocreatepeer=no<BR>>>><BR>>>>
[secondbox]<BR>>>> type=friend<BR>>>> host= <A
href="http://10.0.0.2">10.0.0.2</A><BR>>>>
secret=mysecret<BR>>>><BR>>>> In the second box, my
sip.conf looks like this:<BR>>>><BR>>>>
[general]<BR>>>> allowguest=yes <BR>>>>
autocreatepeer=no<BR>>>><BR>>>>
[secondbox]<BR>>>> type=user<BR>>>> host=<A
href="http://10.0.0.1">10.0.0.1</A><BR>>>>
secret=mysecret<BR>>>><BR>>>> Any ideas on how to
correctly set this up? <BR>>>><BR>>>>
Thanks,<BR>>>> Waldo<BR>>>>
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</A><BR>>><BR>>><BR>>><BR>>>
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