we made available this patch few weeks ago: http://lists.digium.com/pipermail/asterisk-dev/2003-July/001202.html
I hope that It will work with newest chan_sip.c cvs version if not we will update it :)))
It's working for us more than 6 months without any problems.
Best regards MiniTelecom Team
David Harris wrote:
Is it possible to restrict the number of concurrent calls made to a SIP peer? Or maybe the number of concurrent calls made to a particular extension. This way I can avoid asterisk trying to make more voice calls to my remote SIP gateway then I have bandwidth to handle.
/davidh
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