[Asterisk-video] outgoing video-calls don't work
barbara_b02 at libero.it
barbara_b02 at libero.it
Wed Feb 27 09:02:47 CST 2008
Hi all!
Hi Klaus! And thank you for you kindness.
I've made what you said during a SIP->3G call and this is the result:
pc001*CLI> core show channels
Channel Location State Application(Data)
mISDN/1-u28 dialcell at sip_to_cell Up (None)
SIP/101-081dd8c8 03934714462 at default: Up h324m_call(dialcell at sip_to_cell)
2 active channels
1 active call
pc001*CLI> core show channel SIP/101-081dd8c8
-- General --
Name: SIP/101-081dd8c8
Type: SIP
UniqueID: 1204105767.16
Caller ID: 101
Caller ID Name: 101
DNID Digits: 03934714462
State: Up (6)
Rings: 0
NativeFormats: 0x100008 (alaw|h263p)
WriteFormat: 0x2000 (amr)
ReadFormat: 0x2000 (amr)
WriteTranscode: Yes
ReadTranscode: Yes
1st File Descriptor: 28
Frames in: 1933
Frames out: 0
Time to Hangup: 0
Elapsed Time: 0h0m43s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: default
Extension: 03934714462
Priority: 2
Call Group: 0
Pickup Group: 0
Application: h324m_call
Data: dialcell at sip_to_cell
Blocking in: ast_waitfor_nandfds
Variables:
SIPCALLID=YjRlMzlmZDMwYTIzMmVhZmE0ZTY5MGJlMGJkYzRlYTk.
SIPUSERAGENT=X-Lite release 1011s stamp 41150
SIPDOMAIN=192.168.23.21
SIPURI=sip:101 at 192.168.23.244:22836
CDR Variables:
level 1: clid="101" <101>
level 1: src=101
level 1: dst=03934714462
level 1: dcontext=default
level 1: channel=SIP/101-081dd8c8
level 1: lastapp=h324m_call
level 1: lastdata=dialcell at sip_to_cell
level 1: start=2008-02-27 10:49:27
level 1: answer=2008-02-27 10:49:38
level 1: end=2008-02-27 10:49:38
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: accountcode=support
level 1: uniqueid=1204105767.16
pc001*CLI> core show channel mISDN/1-u28
-- General --
Name: mISDN/1-u28
Type: mISDN
UniqueID: 1204105767.17
Caller ID: dialcell
Caller ID Name: (N/A)
DNID Digits: (N/A)
State: Up (6)
Rings: 0
NativeFormats: 0x8 (alaw)
WriteFormat: 0x4 (ulaw)
ReadFormat: 0x4 (ulaw)
WriteTranscode: Yes
ReadTranscode: Yes
1st File Descriptor: 40
Frames in: 1829
Frames out: 1149
Time to Hangup: 0
Elapsed Time: 0h0m30s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: sip_to_cell
Extension: dialcell
Priority: 1
Call Group: 0
Pickup Group: 0
Application: (N/A)
Data: (None)
Blocking in: ast_waitfor_nandfds
Variables:
BRIDGEPEER=Local/dialcell at sip_to_cell-f09e,2
TRANSFERCAPABILITY=SPEECH
DIALEDPEERNUMBER=1/03934714462
CDR Variables:
level 1: clid=101
level 1: src=101
level 1: dst=03934714462
level 1: dcontext=default
level 1: channel=mISDN/0-u22
level 1: start=2008-02-27 10:49:27
level 1: answer=2008-02-27 10:49:38
level 1: end=2008-02-27 10:49:38
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1204105767.19
Moreover Asterisk gives a lot of the sequet messages:
[Feb 27 14:44:55] NOTICE[27572]: rtp.c:1285 ast_rtp_read: Unknown RTP codec 126 received from '192.168.23.245'
Changing in h324m_call (in app_h324m.c) the format AST_FORMAT_ALAW|AST_FORMAT_ULAW to either, ALAW or ULAW, nothing changes...
I don't know what else to do...
Thanks, thanks thanks...
Barbara B.
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