Summary: | ASTERISK-10908: Dont play video on console dial | ||
Reporter: | Dirk-Michael brosig (dmbrosig) | Labels: | |
Date Opened: | 2007-11-28 02:31:56.000-0600 | Date Closed: | 2011-06-07 14:07:20 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_playback |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | If i makeo a testcall with console dial test1@testcalls the audio files from anyvideo will play, but the video part is not sending. The anyvideo.wav, .h263, .h264 exist, if i call test2 with sip-phone all works fine. [macro-testcall] exten => s,1,Set(CALLERID(all)=Asterisk <199>) exten => s,2,Wait(0.5) exten => s,3,Playback(anyvideo) exten => s,4,Hangup [testcalls] exten => test1,1,Dial(SIP/anysip,,tM(mtestcall)S(5)) exten => test2,1,Playback(anyvideo) ****** ADDITIONAL INFORMATION ****** Sip debug show all is fine Capabilities: us - 0x2f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h264), peer - audio=0x28042e (gsm|ulaw|alaw|adpcm|ilbc|h263|h264)/video=0x280000 (h263|h264), combined - 0x28042e (gsm|ulaw|alaw|adpcm|ilbc|h263|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.48:33240 Peer video RTP is at port 192.168.1.48:7558 I have sniffered this with Ethereal: playback is not send any videoframes. Using counterpath x-lite as called sip phone. | ||
Comments: | By: Joshua C. Colp (jcolp) 2007-11-28 09:30:13.000-0600 I don't exactly understand what you mean, can you explain the call flow a bit more? Can you also provide the FULL sip debug and console output? By: Dirk-Michael brosig (dmbrosig) 2007-11-29 02:48:17.000-0600 It works now. The sip channel in the installation was an older one (from 1.4.7.1). By: Joshua C. Colp (jcolp) 2007-11-29 08:02:08.000-0600 Closed as it has already been fixed in latest. |