[Asterisk-Users] problems with eyebeam - video phone

Wed Aug 17 08:31:19 MST 2005

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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