[asterisk-dev] rtp early media handle with buggy sip device

[TSAREGORODTSEV, Yury]

Hello, 
can anyone suggest what to do, 
buggy sip pbx doesn't send 183 Session in Progress at all on any outgoing calls from * to this pbx. 
Its just starting RTP FLOW right after 180 Ringing. 
because of this we have a lot of issues with early media, most of clients can't hear it. 

Any suggestions? 






Sincerely Yours, 
Tsaregorodtsev Yury 
Bridge Communication Billing & Settlement Plan Limited 
Fernhills Business Center, Foerster Chambers, 
Todd Street Bury, Gtr Manchester, BL9 5BJ 
United Kingdom 
Tel: +441570200000 
ICQ: 622719210 
MSN: y.tsaregorodtsev at live.com 
Skype: tsarik-108 
web: www.bridgecommunication.co.uk 

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