search for: h263p

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 s=SIP Call c=IN IP4 192...

..., I use an asterisk with 4 peers all on the same LAN. My sip.conf looks like that: [general] bindport=1628 videosupport=yes limitonpeers = yes sendrpid = yes trustrpid = no disallow=all allow = gsm # 3 allow = alaw # 8 allow = ulaw # 0 allow = g722 # 9 allow = g726 # 111 allow = h263 # 34 allow = h263p # 98 allow = h264 # 99 [peerA] type=friend secret=kokolala nat=no host=dynamic canreinvite=no context=koko dtmfmode=rfc2833 disallow=all allow = gsm allow = alaw allow = ulaw allow = g722 allow = g726 allow = h263 allow = h263p allow = h264 qualify=no <snip more similar peers> For clients...

...H263 for ID 34 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description format h263-1998 for ID 103 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description format H264 for ID 99 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Capabilities: us - 0x3c0002 (gsm|h261|h263|h263p|h264), peer - audio=0x380006 (gsm|ulaw|h263|h263p|h264)/video=0x380000 (h263|h263p|h264), combined - 0x380002 (gsm|h263|h263p|h264) [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-e...

...-- Started music on hold, class 'default', on channel 'PJSIP/7000-00000000' -- Channel PJSIP/7000-00000000 joined 'softmix' base-bridge <52997aa1-eb00-481c-8c56-e26d78d01515> [May 7 16:21:32] WARNING[20789]: channel.c:834 ast_best_codec: Don't know any of (h263|h263p|h264) formats [May 7 16:21:32] WARNING[20789]: channel.c:834 ast_best_codec: Don't know any of (h263|h263p|h264) formats > 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to 192.168.8.203:17200 > 0x7f46f4187280 -- Probation passed - setting RTP source address to 192....

...ines) --- -- Attempting call on SIP/403 for s at broadcast:1 (Retry 1) Video is at 172.16.148.1 port 18182 Audio is at 172.16.148.1 port 18108 Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x80000 (h263) to SDP Adding codec 0x100000 (h263p) to SDP Adding codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 172.16.148.129:36042: INVITE sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901 SIP/2.0 Via: SIP/2.0/UDP 172.16.148.1:5060;branch=z9hG4bK1d87664a;rport From: "555&qu...

...txt -rw-rw---- 1 asterisk asterisk 55724 2011-10-11 08:06 msg0008.wav -rw-rw---- 1 asterisk asterisk 5715 2011-10-11 08:06 msg0008.WAV Codec negotiation: Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0xc (ulaw|alaw)/video=0x380000 (h263|h263p|h264)/text=0x0 (nothing), combined - 0x38000c (ulaw|alaw|h263|h263p|h264) In asterisk.conf we even activate transcode_via_sln = yes ;Build transcode paths via SLINEAR,instead of directly....

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

--- listas iPfone <listas@ipfone.com.br> wrote: > Hi all > > I need to know if the video support for 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

...alnet=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 scrubbed... URL: http://list...

...ox. I did following configuration In Sip.conf videosupport=yes [phone1] type=friend host=dynamic context= employees mailbox=101 at default callerid="phone1<101>" disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h264 allow=h263 [phone2] ?type=friend host=dynamic context= employees mailbox=102 at default callerid="phone2<102>" disallow=all allow=ilbc allow=g723 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263 allow=h263p allow=h261 In extensio...

...;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 : asterisk Pa...

...0@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: ====================================...

...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...

...ID 97 Found audio description format G726-16 for ID 98 Found audio description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event...

...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 feedback is very much appreciated. Cheers, Stephen Bosch

...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> -- PBX --...

...ase. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger <6003> dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI> -- Executing Dial("SIP/6003-94ec", "SIP/6004|60|trm") in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -...

...l 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.digium.com/pipermail/a...

...Asterisk servers setup as a SIP client connected to a 4th Asterisk server. Assuming it is possible, here is the SIP Client SIP.CONF: [general] register => 103:1234 at yy.yy.yy.yy/699 defaultexpirey=1800 maxexpirey=3600 relaxdtmf=yes videosupport=yes disallow=all allow=ulaw allow=gsm allow=h263p canreinvite=no limitonpeer=yes notifyringing=yes notifyhold=yes externip=xx.xx.xx.xx.xx fromdomain=xx.xx.xx.xx localnet=192.168.0.0/255.255.255.0 [yy.yy.yy.yy] type=friend host=yy.yy.yy.yy insecure=port,invite [699] type=friend secret=1234 dial=SIP/699 callerid=Video <699> allowsubscribe=ye...

...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...