[asterisk-dev] Working with SIP Communicator or disabling reINVITE

[Samir S]

Thanks, this does stop Asterisk from sending the reINVITEs...
Does the setting directrtpsetup=yes work ?
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 )

<--- SIP read from x.x.19.205:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.19.216:5060;rport=5060;branch=z9hG4bK061372f3;received=
x.x.19.216
From: "user2" <sip:user2 at x.x.19.216>;tag=as35372950
To: <sip:user1 at x.x.19.205:5060;transport=udp>;tag=a47b8066
Call-ID: 5c1088ad43b03d0a39a06569779b0fd3 at x.x.19.216
CSeq: 102 INVITE
User-Agent: SIP Communicator 1.0 CVS-Tue_Sep_25_18-24-10_IST_2007
Content-Type: application/sdp
Contact: "user1" <sip:user1 at x.x.19.205:5060;transport=udp>
Content-Length: 103

v=0
o=user1 0 0 IN IP4 x.x.19.205
s=-
c=IN IP4 x.x.19.205
t=0 0
m=audio 5000 RTP/AVP 0 3 8

<--- Reliably Transmitting (no NAT) to x.x.19.253:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.19.253
:5060;branch=z9hG4bK6dc9c8d84ff595ea1d1a84d72cb34a43;received=x.x.19.253
From: "user2" <sip:user2 at x.x.19.216>;tag=498d21e1
To: <sip:1 at x.x.19.216>;tag=as7dc3a436
Call-ID: 87055dd8887755e46e6760cd87c7dad7 at 0.0.0.0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1 at x.x.19.216>
Content-Type: application/sdp
Content-Length: 233

v=0
o=root 26946 26946 IN IP4 x.x.19.216
s=session
c=IN IP4 x.x.19.216
t=0 0
m=audio 13886 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv



On 9/25/07, Atis Lezdins <atis at iq-labs.net> wrote:
>
> On Tuesday 25 September 2007 13:35:48 Samir S wrote:
> > Hello All,
> >
> > Does anyone have experience of interworking Asterisk with the SIP
> > communicator ?
> > It seems that the Sip communicator does not support reINVITE
> > Is it possible to disable this generate of reINVITE in Asterisk and make
> it
> > default to a pure proxy (for testing purposes only) ?
>
> Set "canreinvite=no" in sip.conf for corresponding user (maybe it works
> globally, but i'm not sure).
>
> Regards,
> Atis
>
>
> --
> Atis Lezdins
> VoIP Developer,
> IQ Labs Inc.
> atis at iq-labs.net
> Skype: atis.lezdins
> Cell Phone: +371 28806004
> Work phone: +1 800 7502835
>
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