[Sip-implementors] History-Info

2011-12-22 Thread Johnson, Michael A
I am seeking clarification on the History-Info RFC 4424 In a transit scenario, an incoming call that is immediately redirected to another party, I am getting rejected based on the History-Info field. This is only where there is no A-party identity provided on the initial call. On the outgoing in

Re: [Sip-implementors] Re-Invite codec renegotiation.

2011-07-05 Thread Johnson, Michael A
egotiation modified. I appreciate your patience & assistance. -Original Message- From: Worley, Dale R (Dale) [mailto:dwor...@avaya.com] Sent: Saturday, 2 July 2011 5:25 AM To: Johnson, Michael A; sip-implementors@lists.cs.columbia.edu Subject: RE: Re-Invite codec renegotiation. > From: J

Re: [Sip-implementors] Re-Invite codec renegotiation.

2011-06-30 Thread Johnson, Michael A
ssage- From: Worley, Dale R (Dale) [mailto:dwor...@avaya.com] Sent: Friday, 1 July 2011 12:51 AM To: Johnson, Michael A; sip-implementors@lists.cs.columbia.edu Subject: RE: Re-Invite codec renegotiation. ____ From: Johnson, Michael A [michael.a.john...@t

Re: [Sip-implementors] Re-Invite codec renegotiation.

2011-06-29 Thread Johnson, Michael A
v -Original Message- From: Worley, Dale R (Dale) [mailto:dwor...@avaya.com] Sent: Thursday, 30 June 2011 5:58 AM To: Johnson, Michael A; sip-implementors@lists.cs.columbia.edu Subject: RE: Re-Invite codec renegotiation. > Hold invite: > > [re-INVITE without SDP] > > Hold OK

Re: [Sip-implementors] Re-Invite codec renegotiation.

2011-06-29 Thread Johnson, Michael A
I apologise if I have been unclear in the posting. A point to mention is that this vendor (Mitel) do not support 'hold' in the correct sense. They use 'consultation-hold' or the first stage of a transfer as their hold mechanism. While I disagree with this approach, it is not the issue I am figh

[Sip-implementors] Re-Invite codec renegotiation.

2011-06-28 Thread Johnson, Michael A
I have an issue with a vendor where a call placed from a phone that negotiates G.729 (ISP supports both G.729 & G.711) 200 OK with SDP: t=0 0 m=audio 9000 RTP/AVP 8 0 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 When I place the call on-hold,