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2003 Jun 16
2
The same SIP problems...SORRY!
...can help!! Thanks in advance! michelle PD:188.208.12.237 is the asterisk IP (...) s=session c=IN IP4 188.208.12.237 t=0 0 =audio 13532 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 229.159.241.112:5060 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-3a5246f7- 8c6b606-10eb From: ;tag=0-13c4-3a5246f7-8c6b604-c3a To: ;tag=as52ed0a6a Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765@188.208.12.37 CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: Content-Type: application/sdp Content-Length: 135 v=0 o=root 11673 11673 IN IP4 188.208.12.237 s=session c=IN IP4 188.208.12....
2003 Jun 11
4
some sip questions AGAIN
I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have