<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:arial, helvetica, sans-serif;font-size:10pt"><DIV style="FONT-SIZE: 10pt; FONT-FAMILY: arial, helvetica, sans-serif">Thanks, that is a very helpful endorsement.<BR><BR>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">----- Original Message ----<BR>From: Anthony Minessale <anthony.minessale@gmail.com><BR>To: freeswitch-dev@lists.freeswitch.org<BR>Sent: Thursday, May 15, 2008 9:27:56 AM<BR>Subject: Re: [Freeswitch-dev] Sofia-SIP for SIP trunking<BR><BR>Well it was the 5th SIP stack we tried including all the other ones on your list and we have some people doing 400+ calls per second so it's probably a good choice.<BR><BR><BR>
<DIV class=gmail_quote>On Thu, May 15, 2008 at 8:54 AM, Jim Thomas <<A href="mailto:jimthomasembedded@yahoo.com" target=_blank rel=nofollow ymailto="mailto:jimthomasembedded@yahoo.com">jimthomasembedded@yahoo.com</A>> wrote:<BR>
<BLOCKQUOTE class=gmail_quote style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">Hello,<BR>Is Sofia-SIP a good choice to add SIP trunking support to a PBX?<BR>Are there any known limitations or weaknesses relative to alternatives such as sipX, reSIProcate, etc?<BR>Is SIP trunking based on Sofia-SIP interoperable with a broad range of SIP trunking service providers?<BR>Thanks.<BR>Jim<BR><BR><BR><BR><BR><BR><BR>_______________________________________________<BR>Freeswitch-dev mailing list<BR><A href="mailto:Freeswitch-dev@lists.freeswitch.org" target=_blank rel=nofollow ymailto="mailto:Freeswitch-dev@lists.freeswitch.org">Freeswitch-dev@lists.freeswitch.org</A><BR><A href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev" target=_blank rel=nofollow>http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev</A><BR>UNSUBSCRIBE:<A href="http://lists.freeswitch.org/mailman/options/freeswitch-dev"
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