search for: h261

Built asterisk 11.8.1 on a Debian VPS. Testing using ekiga. If h261 is checked in ekiga's video format list I have video, and mouse over the video window shows it to be using h261. But then I get the following lines a dozen or more times in the CLI: [Mar 21 16:25:32] WARNING[31818][C-00000010]: file.c:1241 ast_writefile: No such format 'h261' The pro...

...eign.wav is a .wav file that plays nicely on my 1.4 server. Here is what the console displays: -- Executing [730 at customers:2] Playback("SIP/user_xxx-00000012", "/home/phones/common/moh/moha/Sovereign") in new stack Unable to find a codec translation path from 0x40004 (ulaw|h261) to 0x40 (slin) Unable to open /home/phones/common/moh/moha/Sovereign (format 0x40004 (ulaw| h261)): No such file or directory ast_streamfile failed on SIP/user_xxx-00000012 for /home/phones/common/moh/moha/Sovereign I was under the impression that I didn't have to do anything to get slin s...

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] translate.c: No translator path from unknown to unknown Mar 21 22:02:34 WARNING[2...

...and everything from making this work to H.323 (which is a pain anyway) isn't working out so well. Thanks for any help, Rob >From sip.conf: [general] videosupport=yes [3005] type=friend secret=XXXXXX mailbox=3005 host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=h261 allow=h263 [3007] type=friend regexten=3007 mailbox=3007 secret=XXXXXX host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=h261 allow=h263

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 (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 This is de sip debug on INVITE (Ekiga calls GXV3140) : v=0 o=grandstream 8000 8000 IN IP4 192.168.1.103...

HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam<1102> canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes jitterbuffer=no disallow=all ;allow=ulaw ;allow=alaw allow=h261 allow=h263 allow=gsm -- IAX2/192.168.0.7:4569/2 answered SIP/1102-62b6 Sep 23 11:49:33...

...or h.263 is > active in version stable > 1.0.7 to use with eyeBeam in asterisk it works for me... [2222] type=friend secret=xxxx auth=md5 callerid="myCallerId" <2222> canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

...IONS, BYE, REFER Content-Type: application/sdp Content-Length: 268 v=0 o=root 1049 1049 IN IP4 <asterisk ip> s=session c=IN IP4 <asterisk ip> t=0 0 m=audio 16112 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 18274 RTP/AVP 31 34 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 (NAT) to <gateway2 ip>:5060 -- Called 3004 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b From: "3002" <sip:3002@<asterisk ip>>;tag=as2f4208e5 To: <sip:<gateway2 ip>> Call-ID:...

...at : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 200.93.xxx.xb Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Username : 105 Codecs : 0xc011f (g723|gsm|ulaw|alaw|g726|g729|h261|h263) Codec Order : (g729|g723|gsm|g726|ulaw|alaw|h261|h263) Status : UNKNOWN Useragent : Full Contact : sip:105@192.168.0.253:5060 And this is the output of sip debug peer 105 when I call to *98 (for voice messages): asterisk*CLI> sip debug peer 105 SIP Debugging Enabled for...

Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn

...nf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to /etc/asterisk/sip_custom.conf allow=gsm allow=h261 allow=h263 allow=h263p videosupport=yes _________________________________________________________________ Windows Live?: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 -------------- next part -------------- An HTML attachment was scrub...

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

...efaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk) Auto-Framing : No Status : UNKNOWN Useragent : Reg. Contact : Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Refuse Sess-Refresh : uac Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : as...

...622118752830@157.161.x.x Sending to 157.161.x.x : 5060 (NAT) Found peer 'PBX-in'' Found RTP audio format 8 Peer audio RTP is at port 172.28.32.2:54204 Peer video RTP is at port 172.28.32.2:65535 Found description format PCMA Capabilities: us - 0x1f060e (gsm|ulaw|alaw|speex|ilbc|jpeg|png|h261|h263| h263p), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 4144400xxxx in fromPBX (domain 157.161.x.x) Now what I call an anonymous call: ========================...

...IP4 216.82.224.202 s=session c=IN IP4 216.82.224.202 t=0 0 m=audio 17050 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12426 RTP/AVP 31 34 103 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=sendrecv _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://windowslive.com/Campaign/SocialNetworking?ocid=PID23285::T:WLMTAGL:ON:...

...el-id=428016;max-mbps=35000;max-fs=3600;max-smbps=323500 a=rtpmap:99 H263-1998/90000 a=fmtp:99 custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo a=rtpmap:34 H263/90000 a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200 a=rtpmap:31 H261/90000 a=fmtp:31 cif=1;qcif=1;maxbr=19200 a=rtcp-fb:* nack pli a=sendrecv a=content:main a=label:11 a=answer:full m=application 5078 UDP/BFCP * c=IN IP4 192.168.50.10 a=floorctrl:c-s a=confid:1 a=floorid:2 mstrm:12 a=userid:1 a=setup:passive a=connection:new m=video 48264 RTP/AVP 99 34 31 c=IN IP4 1...

...x200) audio speex (SpeeX) > 1024 (1 << 10) (0x400) audio ilbc (iLBC) > 65536 (1 << 16) (0x10000) image jpeg (JPEG image) > 131072 (1 << 17) (0x20000) image png (PNG image) > 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) What is the 16 bit signed Linear PCM format? How do I get Asterisk (1.2) to use such a sound file instead of a *.gsm file? Your fe...

...ype: SIP UniqueID: 1329515589.179 LinkedID: 1329515589.179 Caller ID: 1064 Caller ID Name: device Connected Line ID: (N/A) Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) Rings: 0 NativeFormats: 0x3c0002 (gsm|h261|h263|h263p|h264) WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 17 Frames in: 153 Frames out: 385 Time to Hangup: 0 Elapsed Time: 0h0m10s Direct Bridge: <none> Indirect Bridge: <none> -- PB...

...ss_alert=8 disallow=all allow=all dtmfmode=inband faststart=yes context=internal [Avaya] type=friend context=internal host=10.1.129.247 port=1720 canreinvite=no disallow=all allow=alaw dtmfmode=inband *Here is also the content of sip_custom.conf:* [general] context=internal videosupport=yes allow=h261 allow=h263 allow=h263p bindaddr=10.1.129.231 srvlookup=yes conreinvitte=no [1000] type=friend secret=malvin123 host=dynamic dtmfmode=inband disallow=all allow=all nat=yes Thanks & regards, Malvin -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists...