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Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site A Site B Site C ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 Thanks -------------- next part -------------- An HTML attachment was scrubbed...

Hello My provider allows to activate/deactivate a forwarding rule by sending a SIP MESSAGE. This is done outside a call. That is, while there is no ongoing call, a SIP client just sends the following message: MESSAGE sip:543951354657 at callfwd.sip.providerx.com SIP/2.0 Call-ID: b9ba106e-613a-46b9-8a4d-0efb4dc0a0f2 CSeq: 1 MESSAGE To: <sip:543951354657 at callfwd.sip.providerx.com> From: "543951354657" <sip:543951354657 at sip.providerx.com>;tag=... Max-Forwards: 70 Supported: 100rel User-Agent: Provide...

Hi All am running asterisk 1.8 , its a realtime asterisk.have recently noticed an issues where am getting loads of these errors SIP/blabla requested media update control 26, passing it to SIP/ProviderX this literally keeps flooding the screen untill the call is answered and then i get normal "locally bridging two channels" infact, this has caused cross-talk (3-way call) between some random calls. infact am not sure whether this error has caused this, i have filtered the logs for a call...