<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>>>> _______________________________________________<br>>>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com
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