[Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite

Tue Jul 1 19:10:41 MST 2003

To find out what version yuor using, dial *999 and a debug/trace window will
appear.  In the SIP messages it will indicate the type of UA your using and
the version.  example below:   try another call attempt with this window
open and capture the call flow and send it to me.  See below in bold or
(User-Agent: X-PRO build 1035).

Mike

SIP: 140.186.105.40:5060
RTP: 140.186.105.40:8000
NAT: 140.186.105.40

PXY#0: 140.186.104.157:5060

SEND >> 140.186.104.157:5060
REGISTER sip:sp01.to-talk.com SIP/2.0
Via: SIP/2.0/UDP 140.186.105.40:5060
From: <sip:700000011 at sp01.to-talk.com>
To: <sip:700000011 at sp01.to-talk.com>
Contact: "watson can" <sip:700000011 at 140.186.105.40:5060>
Call-ID: 729B4EE5D59C4BB088F3EAA762213865 at sp01.to-talk.com
CSeq: 27940 REGISTER
Expires: 500
User-Agent: X-PRO build 1035
Content-Length: 0

RECEIVE << 140.186.104.157:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 140.186.105.40:5060
From: <sip:700000011 at sp01.to-talk.com>
To:
<sip:700000011 at sp01.to-talk.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3fa3
Call-ID: 729B4EE5D59C4BB088F3EAA762213865 at sp01.to-talk.com
CSeq: 27940 REGISTER
Contact: <sip:700000011 at 140.186.105.40:5060>;q=0.00;expires=500
Server: Sip EXpress router (0.8.11pre29 (i386/linux))
Content-Length: 0
Warning: 392 140.186.104.157:5060 "Noisy feedback tells:  pid=2962
req_src_ip=140.186.105.40 req_src_port=5060 in_uri=sip:sp01.to-talk.com
out_uri=sip:sp01.to-talk.com via_cnt==1"

Michael Kane (President/CEO)
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
www.to-talk.com
508-295-2826
----- Original Message ----- 
From: "Moshe Yudkowsky" <speech at pobox.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, July 01, 2003 9:32 PM
Subject: Re: [Asterisk-Users] Today's Message from linphone; update on
Khpone and SJPhone and X-Lite

> At 20:08 2003-07-01 -0400, Michael Kane wrote:
> >What version of X-Lite are you using.  The latest is build v1035.  There
> >where problems in earlier releases with SDP values, that could be the
reason
> >you not seeing invites or media.  I had issues only with the media not
> >setting as X-lite tried to negotiate media with another endpoint and teh
SDP
> >was hosed.
> >
> >Mike
>
>
> Mike,
>
> I downloaded the version I'm using late last week or early this week. It
> ought to be the latest. There's no way to tell from looking at the app
> (that I can find) what build it is.
>
> Let's see... created June 18th. That's pretty recent. I think it may be
> bug-report time.
>
> Any softphones you recommend for PC or for Linux? I'm actually rather
> disappointed with everything I've tested, with the exception of SJPhone
> (but I've only fiddled with it very briefly).
>
>
>
>
> >Michael Kane
> >To-Talk Communications LLC.
> >37 Sandusky Dr.
> >Wareham, Ma. 02571
> >www.to-talk.com
> >508-295-2826
> >----- Original Message -----
> >From: "Moshe Yudkowsky" <speech at pobox.com>
> >To: <asterisk-users at lists.digium.com>
> >Sent: Tuesday, July 01, 2003 7:24 PM
> >Subject: [Asterisk-Users] Today's Message from linphone; update on Khpone
> >and SJPhone and X-Lite
> >
> >
> > > Today's "frustrated programmer" award goes to Linphone, which has the
> > > following debug output:
> > >
> > > > (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip
phone
> >did not answered properly to my sdp offer!
> > >
> > > I get this message when I connect to linphone using a softphone, or
when
> > > I try to use linphone to connect to asterisk and listen to an
> > > announcement. I suspect that this is a linphone problem... other
clients
> > > don't report problems.
> > >
> > > In other news, according to my trace of Ethernet packets, the PC
> > > softphone X-Lite sends no RTP packets -- neither UDP nor TCP -- to my
> > > Linux softphone, nor does it play out the UDP packets that it
receives.
> > > This is not an asterisk problem because the PC's SJphone does work --
> > > sortof.
> > >
> > > The PC's SJPhone does send/receive packets directly to asterisk. But
> > > there seems to be a problem with someone's negotiation protocol --
> > > Kphone seems to expect GSM and SJPhone is apparently sending G.711.
You
> > > can imagine how that sounds. More later if I get it straightened out.
> > >
> > > --
> > >   Moshe Yudkowsky * http://www.Disaggregate.com
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> -- 
>   Moshe Yudkowsky
>   Disaggregate
>   2952 W Fargo
>   Chicago, IL 60645 USA
>
>   www.Disaggregate.com
>   speech at pobox.com
>   +1 773 764 8727
>
> _______________________________________________
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>