[Asterisk-Users] problems with eyebeam - video phone

Tue Aug 16 23:39:12 MST 2005

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                                             
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             m.com                                                      cc 

                                                                   Subject 
             16/08/2005 20.48          RE: [Asterisk-Users] problems with  
                                       eyebeam - video phone               

             Please respond to                                             
              Asterisk Users                                               
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              Non-Commercial                                               
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             <asterisk-users at l                                             
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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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