[Asterisk-video] 3G <-->SIP audio problems
aster vdo
astervdo at gmail.com
Wed Aug 20 22:23:09 CDT 2008
Hi,
I am doing a video call from a 3G to SIP.
The video works fine, but there is no audio for both the parties.
I using Asterisk 1.4.20.1
and i get the following warning message on asterisk
WARNING[15840] chan_sip.c: Asked to transmit frame type 8192, while native
formats is 0x4 (ulaw)(4) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)
and the show channel commands results in the following output.
*CLI> core show channel Zap/1-1
-- General --*CLI>
Name: Zap/1-1
Type: Zap
UniqueID: 1219243635.23
Caller ID: XXXXXXXX
Caller ID Name: (N/A)
DNID Digits: XXXXXXX
State: Up (6)
Rings: 1
NativeFormats: 0x44 (ulaw|slin)
WriteFormat: 0x4 (ulaw)
ReadFormat: 0x4 (ulaw)
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 19
Frames in: 3275
Frames out: 2532
Time to Hangup: 0
Elapsed Time: 0h1m5s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: default
Extension: s
Priority: 2
Call Group: 0
Pickup Group: 0
Application: h324m_gw
Data: dial at cell_to_sip
Blocking in: ast_waitfor_nandfds
Variables:
ul1=65535
CALLEDTON=33
ANI2=0
TRANSFERCAPABILITY=DIGITAL
CDR Variables:LI>
level 1: clid=XXXXXXXX
level 1: src=XXXXXXXX
level 1: dst=s
level 1: dcontext=default
level 1: channel=Zap/1-1
level 1: lastapp=h324m_gw
level 1: lastdata=dial at cell_to_sip
level 1: start=2008-08-20 15:47:15
level 1: answer=2008-08-20 15:47:30
level 1: end=2008-08-20 15:47:30
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1219243635.23
Is there any thing i am doing wrong..
regards
aster
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-video/attachments/20080821/edfffeb9/attachment.htm
More information about the asterisk-video
mailing list