[Asterisk-Users] problems with eyebeam - video phone

Jimmy Smith jimmy.voippro at gmail.com
Wed Aug 17 08:32:15 MST 2005 

Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x80000 formats

is from what you pasted btw..

Don't know any of 0x80000 formats 

is
524288 (1 << 19)  (0x80000)  video    h263   (H.263 Video)

meaning it downst understand it or find it



On 8/17/05, Jimmy Smith <jimmy.voippro at gmail.com> wrote:
> quickly this looks like a incompatible codec.. or unrecognized..
> 
> show codecs on CLI>
> 
> show show
>  262144 (1 << 18)  (0x40000)  video    h261   (H.261 Video)
>     524288 (1 << 19)  (0x80000)  video    h263   (H.263 Video)
>    1048576 (1 << 20) (0x100000)  video   h263p   (H.263+ Video)
> does it ?
> 
> On 8/17/05, asterisk at frameweb.it <asterisk at frameweb.it> wrote:
> > Thank you for your answer.
> > I didn't register on the domain of the Eyebeam software, actually I don't
> > understand how to do that!
> > I bouught 5 eyebeam activation keys and I am trying with the first 2 of
> > them
> >
> > On the Eyebeam side (both eyebeam), I only enabled the "Basic H.263" codec,
> > no other.
> >
> > If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263, the
> > two video phone speak without any problem (but without any video)
> > If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263, the
> > first video phone call the second, the second answer and immediately
> > the call ends.
> >
> > If Ilook at /var/log/asterisk/full, I see:
> > ........
> > Aug 17 08:37:06 VERBOSE[14731]:     -- AGI Script dialparties.agi
> > completed, returning 0
> > Aug 17 08:37:06 VERBOSE[14731]:     -- Executing Dial("SIP/551-eac0",
> > "SIP/552|25|tr") in new stack
> > Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL)
> > Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0
> > Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0
> > Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x80000 formats
> > Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552
> > Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0
> > Aug 17 08:37:06 VERBOSE[14731]:     -- Called 552
> > Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but
> > retaining packet) on '7e677c350356227149bd8469193dae0f at 192.168.69.10'
> > Request 102: Found
> > Aug 17 08:37:06 VERBOSE[14731]:     -- SIP/552-ff46 is ringing
> > Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102
> > Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on
> > '7e677c350356227149bd8469193dae0f at 192.168.69.10' of Request 102: Found
> > Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop:
> > <sip:552 at 192.168.69.122:5060>
> > Aug 17 08:37:10 VERBOSE[14731]:     -- SIP/552-ff46 answered SIP/551-eac0
> > Aug 17 08:37:10 WARNING[14731]: No path to translate from SIP/551-eac0(2)
> > to SIP/552-ff46(524288)
> > Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make
> > SIP/551-eac0 compatible with SIP/552-ff46
> > Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement outUse
> > counter
> >
> >
> > It seems the problem documented in bug
> > http://bugs.digium.com/bug_view_page.php?bug_id=0003709
> > but actually it is not exactly the same.
> >
> > moreover: is there any way to put the patch described in
> > http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in *)
> > in asterisk 1.0.9 and not asterisk CVS HEAD ?
> >
> > Any help will be greatly appreciated.
> >
> > Andrea
> >
> >
> >
> >
> >             "Carlos Alperin"
> >             <calperin at senecac
> >             om.net>                                                    To
> >             Sent by:                  "'Asterisk Users Mailing List -
> >             asterisk-users-bo         Non-Commercial Discussion'"
> >             unces at lists.digiu         <asterisk-users at lists.digium.com>
> >             m.com                                                      cc
> >
> >                                                                   Subject
> >             16/08/2005 20.48          RE: [Asterisk-Users] problems with
> >                                       eyebeam - video phone
> >
> >             Please respond to
> >              Asterisk Users
> >              Mailing List -
> >              Non-Commercial
> >                Discussion
> >             <asterisk-users at l
> >             ists.digium.com>
> >
> >
> >
> >
> >
> >
> > Hi,
> >
> > I get Eyebeam working with an older version of Asterisk 1.0.2(I believe). I
> > only use H.263 and SIP. (G.729)
> >
> > Now, the more important question is if you register on the domain on the
> > Eyebeam software. I found that this was the full secret about this.
> >
> > Let me know your configuration on the Eyebeam side.
> >
> > Regards,
> >
> > Carlos Alperin
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> > pellegrini at frameweb.it
> > Sent: Tuesday, August 16, 2005 11:28 AM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] problems with eyebeam - video phone
> >
> > I am trying to connect two Xten eyeBeam Video Phone
> >
> > No problems in voice connecting.
> >
> > I tryed to modify my sip.conf
> >
> > [general]
> > language=it
> > videosupport=yes
> > ; enable Asterisk video support
> >
> > port = 5060           ; Port to bind to (SIP is 5060)
> > bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
> > disallow=all
> > allow=h263
> > allow=gsm
> > allow=ulaw
> > allow=alaw
> > ; H.263 is our video codec
> > ; allow=h263p
> > ; H.263p is the enhanced video codec
> > context = from-sip-external ; Send unknown SIP callers to this context
> > callerid = Unknown
> >
> > #include sip_nat.conf
> > #include sip_custom.conf
> > #include sip_additional.conf
> >
> > And I left only H.263 basic in codec's configuration in Video Phone.
> > No chance to get the communication in H.263 protocol.
> >
> > I saw that to use H.263+ protocol I need Asterisk CVS.
> > I am not using asterisk CVS
> > I am using asterisk 1.0.9 (last stable version a couple of week ago..)
> >
> > Is there any chance to make asterisk 1.0.9 to support SIP video calls in
> > eyeBeam ?
> >
> > Thanks in advance,
> > Andrea
> >
> > Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.
> >
> > Visitate il sito http://www.frameweb.it
> >
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