[asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID

[jonas kellens]

14:38:01.229941 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length:
889
14:38:01.230127 IP 192.168.4.248.sip > 192.168.4.240.sip: SIP, length:
515
14:38:01.251558 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length:
497
14:38:01.271714 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length:
1060
14:38:01.271904 IP 192.168.4.248.sip > 192.168.4.240.sip: SIP, length:
433
14:38:01.272133 IP 192.168.4.248.sip > 192.168.4.242.sip: SIP, length:
861

is what I see... only SIP, no RTP/UDP...

I guess you're right...

Thank you, Tom.


On Sat, 2009-04-18 at 06:50 -0400, Tom Moore wrote:
> Asterisk still controls the signalling, but the audio path should be
> going through the phones directly.
> Fire up a tcpdump on the Asterisk server to varify this.


> 
> 
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