[Asterisk-Users] SIP Canreinvite
Hi all, Im testing canreinvite = yes in my sip.conf with snom190 and a Atcom320.Atcom320 seems support re-invite, but the snom190? Does anyone known if this phone support it? How I can be sure that it works? Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Canreinvite
Hi all, Im testing canreinvite = yes in my sip.conf with snom190 and a Atcom320.Atcom320 seems support re-invite, but the snom190? Does anyone known if this phone support it? How I can be sure that it works? Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP CANREINVITE
I have a number of internal SIP phones and a number of external SIP clients with the server running Asterisk on the boundary between the two. ie the server has two network cards with an internal private address and an external public address. For security reasons no routing is allowed between the two thus no internal phone can talk directly to external phone or visa versa. I am very happy with this arrangement but would like to reduce the load on Asterisk where possible by allowing reinvites. Obviously only internal phones can talk with each other and external phones can talk directly with other external phones. The problem is the logic in Asterisk doesn't allow this. Ideally Asterisk would look at both sides of a call that is has setup and realise that both were internal and thus could be potentially reinvited or that both are external and could thus be reinvited. Where the source or destination was not on the same network then reinvites would not be possible. Of course everyone's network is different and this may not work for some, but wouldn't this be a good idea? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP canreinvite=yes Broke?
So I have many Cisco 7960's that are running the latest 5.1 Cisco SIP code and I cannot get the phones to talk/RTP to each other. jtodd has had this problem in the past with the 186's. Just wondering if anyone has a reason why Cisco sometimes poop on reinvite is the Cisco code broke? if so we can push on Cisco to fix it. the U is a MAJOR Cisco shop so we have some puhs there. if its * code, I will offer up anything (within reason) to work this out. This prob would be a major issue in rolling * out further if every call HAS to go thru the * server fro bridging. Do other SIP hardsets have this problem? sniff you calls to another SIP hardset and check ot see if RTP is coming from the * server of the other phone? Thanks for any info I can get on this Dave P ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
