Rainer Piper
2014-May-07 05:00 UTC
[asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem
Hi!
my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more.
I tried every combination. silent on both sides.
I compiled pjsip with no resample in pjsip.
./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample*
--disable-video --disable-opencore-amr
is there a way to force asterisk back to do the codec translation?
Attachment:
sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the
B-Leg 7000 NativeFormats: (alaw)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
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server*CLI> core show channel PJSIP/7000-00000001
-- General --
Name: PJSIP/7000-00000001
Type: PJSIP
UniqueID: 1399382022.1
LinkedID: 1399382022.0
Caller ID: 7000
Caller ID Name: (N/A)
Connected Line ID: 7001
Connected Line ID Name: 7001
Eff. Connected Line ID: 7001
Eff. Connected Line ID Name: 7001
DNID Digits: (N/A)
Language: de
State: Up (6)
NativeFormats: (alaw)
WriteFormat: g722
ReadFormat: g722
WriteTranscode: Yes (g722)->(slin)->(alaw)
ReadTranscode: Yes (alaw)->(slin)->(g722)
Time to Hangup: 0
Elapsed Time: 0h3m24s
Bridge ID: 0cbdb7d7-81ed-4c5f-896f-cea37599b148
-- PBX --
Context: outgoing-kamailio
Extension:pjsi
Priority: 1
Call Group: 0
Pickup Group: 0
Application: AppDial
Data: (Outgoing Line)
Call Identifer: [C-00000000]
Variables:
BRIDGEPEER=PJSIP/7001-00000000
DIALEDPEERNUMBER=7000
CDR Variables:
level 1: calledsubaddrlevel 1: callingsubaddrlevel 1: dnidlevel 1:
clid="" <700
level 1: src=7000
level 1: dcontext=outgoin
level 1: channel=PJSIP/7
level 1: lastapp=AppDial
level 1: lastdata=(Outgoi
level 1: start=1399382
level 1: answer=1399382
level 1: end=1399382
level 1: duration=1
level 1: billsec=0
level 1: disposition=8
level 1: amaflags=3
level 1: uniqueid=1399382
level 1: linkedid=1399382
level 1: sequence=1
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server*CLI> core show channel PJSIP/7001-00000000
-- General --
Name: PJSIP/7001-00000000
Type: PJSIP
UniqueID: 1399382022.0
LinkedID: 1399382022.0
Caller ID: 7001
Caller ID Name: 7001
Connected Line ID: (N/A)
Connected Line ID Name: (N/A)
Eff. Connected Line ID: (N/A)
Eff. Connected Line ID Name: (N/A)
DNID Digits: (N/A)
Language: de
State: Up (6)
NativeFormats: (g722)
WriteFormat: g722
ReadFormat: g722
WriteTranscode: No
ReadTranscode: No
Time to Hangup: 0
Elapsed Time: 0h3m51s
Bridge ID: 0cbdb7d7-81ed-4c5f-896f-cea37599b148
-- PBX --
Context: outgoing-kamailio
Extension: 7000
Priority: 1
Call Group: 0
Pickup Group: 0
Application: Dial
Data: PJSIP/7000
Call Identifer: [C-00000000]
Variables:
BRIDGEPEER=PJSIP/7000-00000001
DIALEDPEERNUMBER=7000
DIALEDPEERNAME=PJSIP/7000-00000001
DIALSTATUS=ANSWER
DIALEDTIMEANSWEREDTIME CDR Variables:
level 1: calledsubaddrlevel 1: callingsubaddrlevel 1: dnidlevel 1:
clid="7001"
level 1: src=7001
level 1: dst=7000
level 1: dcontext=outgoin
level 1: channel=PJSIP/7
level 1: dstchannel=PJSIP/7
level 1: lastapp=Dial
level 1: lastdata=PJSIP/7
level 1: start=1399382
level 1: answer=1399382
level 1: end=0.00000
level 1: duration=230
level 1: billsec=228
level 1: disposition=8
level 1: amaflags=3
level 1: uniqueid=1399382
level 1: linkedid=1399382
level 1: sequence=0
Rainer Piper
2014-May-07 05:11 UTC
[asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem
PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper:> Hi! > > my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any > more. I tried every combination. silent on both sides. > > I compiled pjsip with no resample in pjsip. > ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr > is there a way to force asterisk back to do the codec translation? > > Attachment: > sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to > the B-Leg 7000 NativeFormats: (alaw) > > > -- > *Rainer Piper* > Integration engineer > Koeslinstr. 56 > 53123 BONN > GERMANY > > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > >-- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de <http://www.soho-piper.de> ------------------------------------------------------------------------ ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140507/ccb608ac/attachment.html>