Thanks, this does stop Asterisk from sending the reINVITEs...<br>Does the setting directrtpsetup=yes work ? <br>I have enabled it..but still Asterisk (running on x.x.19.216) seems to be sending its own IP ....as shown in the following example for the incoming(
x.x.19.205 in the sdp) and outgoing SIP 200 Ok (x.x.19.216 in the sdp )<br><br><--- SIP read from x.x.19.205:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP x.x.19.216:5060;rport=5060;branch=z9hG4bK061372f3;received=
x.x.19.216<br>From: "user2" <<a href="mailto:sip:user2@x.x.19.216">sip:user2@x.x.19.216</a>>;tag=as35372950<br>To: <sip:user1@x.x.19.205:5060;transport=udp>;tag=a47b8066<br>Call-ID: <a href="mailto:5c1088ad43b03d0a39a06569779b0fd3@x.x.19.216">
5c1088ad43b03d0a39a06569779b0fd3@x.x.19.216</a><br>CSeq: 102 INVITE<br>User-Agent: SIP Communicator 1.0 CVS-Tue_Sep_25_18-24-10_IST_2007<br>Content-Type: application/sdp<br>Contact: "user1" <sip:user1@x.x.19.205
:5060;transport=udp><br>Content-Length: 103<br><br>v=0<br>o=user1 0 0 IN IP4 x.x.19.205<br>s=-<br>c=IN IP4 x.x.19.205<br>t=0 0<br>m=audio 5000 RTP/AVP 0 3 8<br><br><--- Reliably Transmitting (no NAT) to x.x.19.253:5060 --->
<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP x.x.19.253:5060;branch=z9hG4bK6dc9c8d84ff595ea1d1a84d72cb34a43;received=x.x.19.253<br>From: "user2" <<a href="mailto:sip:user2@x.x.19.216">sip:user2@x.x.19.216</a>>;tag=498d21e1
<br>To: <<a href="mailto:sip:1@x.x.19.216">sip:1@x.x.19.216</a>>;tag=as7dc3a436<br>Call-ID: <a href="mailto:87055dd8887755e46e6760cd87c7dad7@0.0.0.0">87055dd8887755e46e6760cd87c7dad7@0.0.0.0</a><br>CSeq: 2 INVITE<br>
User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <<a href="mailto:sip:1@x.x.19.216">sip:1@x.x.19.216</a>><br>Content-Type: application/sdp
<br>Content-Length: 233<br><br>v=0<br>o=root 26946 26946 IN IP4 x.x.19.216<br>s=session<br>c=IN IP4 x.x.19.216<br>t=0 0<br>m=audio 13886 RTP/AVP 3 0 8<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000
<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br><br><br><div><span class="gmail_quote">On 9/25/07, <b class="gmail_sendername">Atis Lezdins</b> <<a href="mailto:atis@iq-labs.net">atis@iq-labs.net</a>> wrote:
</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">On Tuesday 25 September 2007 13:35:48 Samir S wrote:<br>> Hello All,<br>><br>> Does anyone have experience of interworking Asterisk with the SIP
<br>> communicator ?<br>> It seems that the Sip communicator does not support reINVITE<br>> Is it possible to disable this generate of reINVITE in Asterisk and make it<br>> default to a pure proxy (for testing purposes only) ?
<br><br>Set "canreinvite=no" in sip.conf for corresponding user (maybe it works<br>globally, but i'm not sure).<br><br>Regards,<br>Atis<br><br><br>--<br>Atis Lezdins<br>VoIP Developer,<br>IQ Labs Inc.<br><a href="mailto:atis@iq-labs.net">
atis@iq-labs.net</a><br>Skype: atis.lezdins<br>Cell Phone: +371 28806004<br>Work phone: +1 800 7502835<br></blockquote></div><br>