<HTML><BODY style="word-wrap: break-word; -khtml-nbsp-mode: space; -khtml-line-break: after-white-space; ">I tried G711 and GSM and in both cases call quality degraded when the softphone was conferencing more than 2 people (note: not a meetme room).<DIV><BR class="khtml-block-placeholder"></DIV><DIV>- Waldo</DIV><DIV><BR><DIV><DIV>On Dec 7, 2005, at 5:45 AM, xcel wrote:</DIV><BR class="Apple-interchange-newline"><BLOCKQUOTE type="cite"> <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>>>> _______________________________________________<BR>>>> --Bandwidth and Colocation provided by <A href="http://Easynews.com">Easynews.com </A>--<BR>>>><BR>>>> Asterisk-Users mailing list<BR>>>> To UNSUBSCRIBE or update options visit:<BR>>>> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users </A><BR>>><BR>>><BR>>><BR>>> _______________________________________________<BR>>> --Bandwidth and Colocation provided by <A href="http://Easynews.com">Easynews.com</A> --<BR>>><BR>>> Asterisk-Users mailing list <BR>>> To UNSUBSCRIBE or update options visit:<BR>>> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR>><BR><BR>_______________________________________________ <BR>--Bandwidth and Colocation provided by <A href="http://Easynews.com">Easynews.com</A> --<BR><BR>Asterisk-Users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users"> http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR></BLOCKQUOTE></DIV><BR></BLOCKQUOTE><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">_______________________________________________</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">--Bandwidth and Colocation provided by Easynews.com --</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><BR></DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Asterisk-Users mailing list</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">To UNSUBSCRIBE or update options visit:</DIV><DIV style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><SPAN class="Apple-converted-space"> </SPAN><A href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A></DIV> </BLOCKQUOTE></DIV><BR></DIV></BODY></HTML>