similar to: H263-2000 video format

So I was thinking there's no need for a new codec. Am I right? Cheers, K ------=_Part_115139_19855101.1183563060148 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline <div>I&#39;m trying to connect my asterisk 1.4.6 to a system that provides video content (through SIP).</div> <div>Problem is my video system only

By browsing on the mailing list I learned that its possible to generate .h263 asterisk friendly files with gstreamer. The script below it's supposed to do just that, however I get error when trying it out locally. gst-launch filesrc location=AstriDevCon_Europe_2006.mov ! qtdemux name=demux ! ffdec_h263 ! videoscale ! video/x-raw-yuv,width=352,height=288 ! ffenc_h263 rtp-payload-size=512 !

I've tried to put a call file to /var/spool/asterisk/outgoing/ to make an outgoing video call, but not succeeded. I could hear the audio, but no video. The asterisk version is 1.4.10, with videosupport=yes The client is eyebeam 1.5.7, with h263 support. Here are some debug messages. It shows the client and asterisk negotiated the video capabilities without problem. However, the 'show

The Asterisk Development Team has announced the release of Asterisk 11.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.10.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.10.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 12.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 12.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

Hello list, what is the difference between these 2 codecs ? What codec to choose if bandwith is an issue ? (like in most cases I guess) Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101208/d611534a/attachment.htm

The Asterisk Development Team has announced the release of Asterisk 1.8.28.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.28.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 1.8.28.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.28.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hello All, I have a media problem while using sip communicator user agent with asterisk behind NAT.I had enabled the debug mode in asterisk and capture the results.I have attached the results with this mail.Can any one help me to fix the problem? Thanks in advance, Partha __________________________________ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out!

Hello list, I'm trying to set up a video call from my Ekiga client to a Grandstream GXV3140 IP-phone. The call succeeds but there is no video. I have in sip.conf : videosupport=yes disallow=all allow=alaw allow=g726 allow=g729 allow=gsm allow=h261 allow=h263 allow=h263p allow=h264 The Grandstream peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 The Ekiga peer has codecs

Hello All, I'am doing a video call between two Video Phones, and i see that Asterisk is stripping the fmtp parameters for the h263 video line in SDP. For example a line similar to the below is stripped, a=fmtp:xx CIF=4;QCIF=2;F=1;K=1 Asterisk is configured NOT to be present in the Media path (My version : Asterisk 1.4.19.1 ). I have the following enabled in my

Hi, I've a problem configuring my Asterisk. What I try to reach is to interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP) with 1 constraint I can't change : "every RTP flow needs to pass THROUGH Asterisk, and are NOT nated" What I observe : - a call made from a SIP Phone registred in Asterisk to Tandberg works (voice and video bidirectionnal) - a call

Hi, we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One GrandStream GXV3000 is used for the tests. He is registered to asterisk 1.6.2.18 asa well as 1.4.35. Calling echo test is OK on both servers, get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP trunk from both others servers is also working well. What fail, is video on echo test from asterisk 1.4.42

A previous poster mentioned the same thing, with no response: http://lists.digium.com/pipermail/asterisk-users/2004- December/080161.html Fresh asterisk 1.0.5 install on FC3, started with "make samples", nothing fancy. It's so bland, I'm surprised the list isn't full of people having the same trouble. I have several Uniden UIP200 phones and a single Grandstream BudgetTone

Hello, I'm trying to set the Asterisk to leave a video message to the mailbox, but there is some compatibility problem, although h263 is identified as the matching codec, as you can see in the debug messages below: Capabilities: us - 0x80100 (g729|h263), peer - audio=0x43f (g723|gsm|ulaw|alaw|g726|adpcm|ilbc)/video=0xc0000 (h261|h263), combined - 0x80000 (h263) Non-codec capabilities: us -

Hi all, I tried making a video SIP call using Asterisk .... But it didnt work....only voice call works? Regards Faraj Khasib

Hi, Recently we got a new feature request from our customer, they want a report to list the duration that agents putting customer on hold, they want to base on this to measure the agents performance. I cannot find any events in cdr, message logs, or manager interface, only when I enable sip debug, then I can see the ReInvite Event in the cli , some thing like the attached logs, is there any

Hello all, I am trying to leave a video voicemail but am unable to do so. I am using Ekiga (formerly Gnomemeeting) to make a SIP connection to Asterisk 1.2.4. Ekiga supports h261 for video. The call connects and negotiation seems okay. When I leave a message, however, only the audio is recorded. Looking in the log file afterwards I see many messages like this: Mar 21 22:02:34 WARNING[2418]