Hello
you mean while placing a video call ? What info am I looking for in the
debug output ?
Kind regards.
J.
On 21-04-17 12:28, Marcelo Terres wrote:
Did you try to activate DEBUG and set the verbosity to a higher level
(100?) to check what Asterisk tells you about?
Regards,
Marcelo H. Terres <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 20 April 2017 at 12:42, Jonas Kellens <jonas.kell...@telenet.be> wrote:
Hello
in sip.conf I have ;
videosupport=yes
Kind regards.
J.
On 20-04-17 13:09, Marcelo Terres wrote:
I suppose that you enable the video support on sip.conf, right?
Regards,
Marcelo H. Terres <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 19 April 2017 at 13:18, Jonas Kellens <jonas.kell...@telenet.be> wrote:
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME
DESCRIPTION
-----------------------------------------------------------------------------------
1 (1 << 0) (0x1) audio g723
(G.723.1)
2 (1 << 1) (0x2) audio gsm
(GSM)
4 (1 << 2) (0x4) audio ulaw
(G.711 u-law)
8 (1 << 3) (0x8) audio alaw
(G.711 A-law)
16 (1 << 4) (0x10) audio g726aal2
(G.726 AAL2)
32 (1 << 5) (0x20) audio adpcm
(ADPCM)
64 (1 << 6) (0x40) audio slin
(16
bit Signed Linear PCM)
128 (1 << 7) (0x80) audio lpc10
(LPC10)
256 (1 << 8) (0x100) audio g729
(G.729A)
512 (1 << 9) (0x200) audio speex
(SpeeX)
1024 (1 << 10) (0x400) audio ilbc
(iLBC)
2048 (1 << 11) (0x800) audio g726
(G.726 RFC3551)
4096 (1 << 12) (0x1000) audio g722
(G722)
8192 (1 << 13) (0x2000) audio siren7
(ITU
G.722.1 (Siren7, licensed from Polycom))
16384 (1 << 14) (0x4000) audio siren14
(ITU
G.722.1 Annex C, (Siren14, licensed from Polycom))
32768 (1 << 15) (0x8000) audio slin16
(16
bit Signed Linear PCM (16kHz))
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)
2097152 (1 << 21) (0x200000) video h264
(H.264 Video)
4194304 (1 << 22) (0x400000) video mpeg4
(MPEG4 Video)
8388608 (1 << 23) (0x800000) video unknown
(unknown)
16777216 (1 << 24) (0x1000000) video unknown
(unknown)
33554432 (1 << 25) (0x2000000) text unknown
(unknown)
67108864 (1 << 26) (0x4000000) text red
(T.140 Realtime Text with redundancy)
134217728 (1 << 27) (0x8000000) text t140
(Passthrough T.140 Realtime Text)
268435456 (1 << 28) (0x10000000) text unknown
(unknown)
536870912 (1 << 29) (0x20000000) text unknown
(unknown)
1073741824 (1 << 30) (0x40000000) (unk) unknown
(unknown)
2147483648 (1 << 31) (0x80000000) (unk) unknown
(unknown)
4294967296 (1 << 32) (0x100000000) audio g719
(ITU
G.719)
8589934592 (1 << 33) (0x200000000) audio speex16
(SpeeX 16khz)
17179869184 (1 << 34) (0x400000000) audio unknown
(unknown)
34359738368 (1 << 35) (0x800000000) audio unknown
(unknown)
68719476736 (1 << 36) (0x1000000000) audio unknown
(unknown)
137438953472 (1 << 37) (0x2000000000) audio unknown
(unknown)
274877906944 (1 << 38) (0x4000000000) audio unknown
(unknown)
549755813888 (1 << 39) (0x8000000000) audio unknown
(unknown)
1099511627776 (1 << 40) (0x10000000000) audio unknown
(unknown)
2199023255552 (1 << 41) (0x20000000000) audio unknown
(unknown)
4398046511104 (1 << 42) (0x40000000000) audio unknown
(unknown)
8796093022208 (1 << 43) (0x80000000000) audio unknown
(unknown)
17592186044416 (1 << 44) (0x100000000000) audio unknown
(unknown)
35184372088832 (1 << 45) (0x200000000000) audio unknown
(unknown)
70368744177664 (1 << 46) (0x400000000000) audio unknown
(unknown)
140737488355328 (1 << 47) (0x800000000000) audio testlaw
(G.711 test-law)
281474976710656 (1 << 48) (0x1000000000000) video unknown
(unknown)
562949953421312 (1 << 49) (0x2000000000000) video unknown
(unknown)
1125899906842624 (1 << 50) (0x4000000000000) video unknown
(unknown)
2251799813685248 (1 << 51) (0x8000000000000) video unknown
(unknown)
4503599627370496 (1 << 52) (0x10000000000000) video unknown
(unknown)
9007199254740992 (1 << 53) (0x20000000000000) video unknown
(unknown)
18014398509481984 (1 << 54) (0x40000000000000) video unknown
(unknown)
36028797018963968 (1 << 55) (0x80000000000000) video unknown
(unknown)
72057594037927936 (1 << 56) (0x100000000000000) video unknown
(unknown)
144115188075855872 (1 << 57) (0x200000000000000) video unknown
(unknown)
288230376151711744 (1 << 58) (0x400000000000000) video unknown
(unknown)
576460752303423488 (1 << 59) (0x800000000000000) video unknown
(unknown)
1152921504606846976 (1 << 60) (0x1000000000000000) video unknown
(unknown)
2305843009213693952 (1 << 61) (0x2000000000000000) video unknown
(unknown)
4611686018427387904 (1 << 62) (0x4000000000000000) video unknown
(unknown)
My 2 video SIP peers :
sip*CLI> sip show peer 660081914
* Name : 660081914
Realtime peer: Yes, cached
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-660081
Subscr.Cont. : <Not set>
Language : nl
Accountcode : 660081intern
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest : default
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 124
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : Yes
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 1.1.1.1:62850
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 660081914
SIP Options : (none)
Codecs : 0x20010a (gsm|alaw|g729|h264)
Codec Order : (alaw:20,g729:20,gsm:20)
Auto-Framing : No
Status : OK (16 ms)
Useragent : MicroSIP/3.15.1
Reg. Contact : sip:660081914@192.168.1.104:62850;ob
Qualify Freq : 120000 ms
Sess-Timers : Refuse
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
sip*CLI> sip show peer 660081915
* Name : 660081915
Realtime peer: Yes, cached
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-660081
Subscr.Cont. : <Not set>
Language : nl
Accountcode : 660081intern
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest : default
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 124
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : Yes
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 1.1.1.1:62812
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 660081915
SIP Options : (none)
Codecs : 0x20010a (gsm|alaw|g729|h264)
Codec Order : (alaw:20,g729:20,gsm:20)
Auto-Framing : No
Status : OK (16 ms)
Useragent : MicroSIP/3.15.1
Reg. Contact : sip:660081915@192.168.1.105:62812;ob
Qualify Freq : 120000 ms
Sess-Timers : Refuse
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
Kind regards
Jonas
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Check out the new Asterisk community forum at:
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New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
