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I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080213/1f05009a/attachment.htm

...;ve tried the 'G' option in the Dial() application but that splits the call as soon as it is answered, whereas, I need to split the call after it is established based on a DTMF stimulus. Are there any other ways of accomplishing this goal? Any thoughts, ideas? Thank you, Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org

Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? Thanks. -- Rgds, -- Rgds, Hans Yin Web: homeofhans.homeip.net Email: hansyin at gmail.com MSN: hansyin at hotmail.com Skype: hans_yin_vancouver

Asterisk supports SIP-T? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090217/7cc67800/attachment.htm

I am working on a project that requires shared extension. Where shared line looks at the status of a line/trunk, shared extension would look at a series of channels as the same "extension". The users would like to add destination channels on the fly, to provide roaming extensions, but maintaining fixed channels as well. If a call comes in on an extension, the system needs to honor the

What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's

I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

...ut. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070822/894e1b 87/attachment-0001.htm ------------------------------ Message: 14 Date: Wed, 22 Aug 2007 06:23:36 -0400 From: "Raj Jain" <rj2807 at gmail.com> Subject: Re: [asterisk-users] rfc3680, reginfo+xml To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <1971b0b60708220323xfd77a2fgf1622b2a507d7c50 at mail.gmail.com> Content-Type: text/plain; charse...