Displaying 20 results from an estimated 20 matches for "writetranscod".
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writetranscode
2017 Nov 22
3
Chan Local, Originate and slin
...to me - I run Originate command from dialplan:
same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum})
and I get crazy sound distortion in the conference, and I see that
transcoding takes place here:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin at 8000)->(slin at 192000)
ReadTranscode: No
When I do the same from a call file like:
same => n,System(printf "Action: Originate\nActionID: 1\nChannel:
Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" >
/var/spool/asterisk/outgoing/${numbe...
2017 Nov 22
2
Chan Local, Originate and slin
Again - when Originate is run from dialplan, i get:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin at 8000)->(slin at 192000)
ReadTranscode: No
When it's made with a call file (no matter how a call file is created), I
see
NativeFormats: (slin)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: No
ReadTranscode: No
Please note - I do not use any manager API
Can...
2008 Aug 09
1
how to know what codec is being used
Hi,
how would i know what codec is being utilized? currently i have set allow=ilbc disallow=all.
i unset all codecs on x-lite except ilbc.
i tried to make a call and look at the channel i see these. does this mean it is using ulaw? how about writetranscode? does that mean there is no transcoding happening on the call? call is going thru, rtp is also going thru. what i would like to know is does it really use ilbc? i'm using 1.4.18.1. thank you
core show channel SIP/100009-082367b
? NativeFormats: 0x4 (ulaw)
??? WriteFormat: 0x4 (ulaw)
???? Rea...
2016 Jun 30
3
how to join 2 channels using AGI/AMI
...Name: SIP/pbx2-000004b9
Type: SIP
UniqueID: 1467323106.1275
Caller ID: xxxx
Caller ID Name: xxxx
DNID Digits: yyyy
Language: en
State: Up (6)
Rings: 0
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x4 (ulaw)
ReadFormat: 0x4 (ulaw)
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 29
Frames in: 325
Frames out: 44
Time to Hangup: 0
Elapsed Time: 0h0m6s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: c_Queues
Extension: 01
Priority: 1
Call Group: 0
Pick...
2014 Apr 29
1
"CBAnn" channel not going away in Asterisk 12
...r ID: (N/A)
Caller ID Name: (N/A)
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: en
State: Up (6)
NativeFormats: (nothing)
WriteFormat: unknown
ReadFormat: unknown
WriteTranscode: No
ReadTranscode: No
Time to Hangup: 0
Elapsed Time: 0h1m3s
Bridge ID: (Not bridged)
-- PBX --
Context: default
Extension: s
Priority: 1
Call Group: 0
Pickup Group: 0
Application: (N/A)
Data: (Empty)
Call Identifer: (None)
Vari...
2015 Sep 30
3
Change Asterisk MulticastRTP codec
...nnel MulticastRTP/0x7f7........' to get lots of helpful information.
I have noticed that when I do a MULTICAST page and send data from MP3Player, I get no sound on my speakers and get the following from 'core show channel PJSIP/xxx':
NativeFormats: (slin)WriteFormat: slinReadFormat: slinWriteTranscode: No ReadTranscode: No
I have noticed that when I do a UNICAST page and send data from MP3Player, everything works flawlessly and I get the following from 'core show channel MulticastRTP':
NativeFormats: (ulaw)WriteFormat: slinReadFormat: slinWriteTranscode: Yes (slin at 8000)->(ulaw at...
2009 Mar 26
3
Know who's logged in
...-- General --
Name: SIP/303-b2f1c368
Type: SIP
UniqueID: 1238094839.425549
Caller ID: 303
Caller ID Name: Ext. 303
DNID Digits: 7700
State: Up (6)
Rings: 0
NativeFormats: 0x2 (gsm)
WriteFormat: 0x2 (gsm)
ReadFormat: 0x2 (gsm)
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 111
Frames in: 6199
Frames out: 4847
Time to Hangup: 0
Elapsed Time: 3h29m16s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: XXXXXXXXXXX
Extension: XXXXX
Priority: XXXXXX
Call...
2014 May 07
1
asterisk12.2.0 PJSIP2.2.0 codec translation problem
....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
Pic...
2012 Feb 17
0
Calling SIP extension through Local/XXXX negotiates slin codec instead of gsm
...Caller ID: 1064
Caller ID Name: device
Connected Line ID: (N/A)
Connected Line ID Name: (N/A)
DNID Digits: (N/A)
Language: en
State: Up (6)
Rings: 0
NativeFormats: 0x3c0002 (gsm|h261|h263|h263p|h264)
WriteFormat: 0x2 (gsm)
ReadFormat: 0x2 (gsm)
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 17
Frames in: 153
Frames out: 385
Time to Hangup: 0
Elapsed Time: 0h0m10s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: from-internal
Extension:
Priority: 1
Call G...
2017 Nov 22
2
Chan Local, Originate and slin
On Wed, 22 Nov 2017, Dmitriy Serov wrote:
> ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" >
> /var/spool/asterisk/outgoing/${number}-${confnum})
I get:
Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/...
Unknown keyword 'ActionID' at line 2 of
2014 Sep 23
1
Change codec when dial from SIP to DAHDI
...09-23 21:18:53] NOTICE[11634][C-0000000d] chan_sip.c: Changing
codec to 'alaw' for this call because of ${SIP_CODEC} variable
if I check channel with "core show channel xxxxx", got DAHDI/SIP
legs final like this:
NativeFormats: (alaw)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: Yes (slin)->(alaw)
ReadTranscode: Yes (alaw)->(slin)
although two legs finally use alaw both, but transcode use slin in
the middle. is it possible to prevent the transcode?
if that is not possible, then maybe I should give up using G722 as
the preffered codec of ip phone. back to G7...
2016 Jun 30
2
how to join 2 channels using AGI/AMI
this is the point, and the strange thing:
DTMF is set to rfc2833, but is working both on incoming and outgoing calls,
it is not working only on calls generated with the Originate AMI command,
or with the queue member that point to Local dialplan, as you suggested
2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>:
> Looking at your logs it looks like you may need to
2015 Jul 06
0
Unisteam not showing callerid
.../A)
ConnectedLineIDName=(N/A)
DNIDDigits= (N/A)
RDNIS= (N/A)
Parkinglot=
Language= en
State= Ring (4)
Rings= 0
NativeFormat= (ulaw|alaw)
WriteFormat= ulaw
ReadFormat= ulaw
RawWriteFormat= ulaw
RawReadFormat= ulaw
WriteTranscode= No
ReadTranscode= No
1stFileDescriptor= 1652
Framesin= 0
Framesout= 0
TimetoHangup= 0
ElapsedTime= 0h0m0s
BridgeID= (Not bridged)
Context= office
Extension= 4203
Priority= 1
CallGroup= 4
PickupGroup=...
2010 Jul 31
0
MeetMe transcode / format problem
...y wav files are alaw encoded and i allow only alaw codec.
But sometimes the WriteFormat is slin and if i recall the same number the WriteFormat is alaw for the Channel.
Why the channel has sometimes slin and sometimes alaw?
NativeFormats: 0x8 (alaw)
WriteFormat: 0x40 (slin)
ReadFormat: 0x8 (alaw)
WriteTranscode: Yes
ReadTranscode: No
After this I'm going into a conference Room and the Format completely change to slin.
NativeFormats: 0x8 (alaw)
WriteFormat: 0x40 (slin)
ReadFormat: 0x40 (slin)
How can i change the Format for Meetme to alaw which is the NativeFormat.
Thanks for your help.
Daniel
P...
2011 Apr 14
0
Followme() and variables
...Type: SIP
UniqueID: 1302818206.232
LinkedID: 1302818171.226
Caller ID: (N/A)
Caller ID Name: (N/A)
DNID Digits: (N/A)
Language: en
State: Up (6)
Rings: 0
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x4 (ulaw)
ReadFormat: 0x4 (ulaw)
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 48
Frames in: 456
Frames out: 1381
Time to Hangup: 0
Elapsed Time: 0h0m29s
...
Variables:
OUTCID is missing here, which is set in the sip.conf of the originating
user/extension
Thanks for any ideas to solve this.
- Jared
-...
2023 May 05
0
Calls running forever / CDRs inaccurate
...xxxx
Caller ID Name: (N/A)
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: 49xxxxxxxxx
Language: en
State: Ring (4)
NativeFormats: (alaw)
WriteFormat: alaw
ReadFormat: alaw
WriteTranscode: No
ReadTranscode: No
Time to Hangup: 0
Elapsed Time: 13h4m11s
Bridge ID: (Not bridged)
-- PBX --
Context: customer-voipin
Extension: 49xxxxxxxxx
Priority: 26
Call Group: 0
Pickup Group: 0
Application: Dial
Data: SIP/+49xxxxx...
2009 Feb 25
1
Stuck Parked Calls?
I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and
2014 Mar 11
1
PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
...m= (N/A)
ConnectedLineIDName=(N/A)
DNIDDigits= (N/A)
RDNIS= (N/A)
Parkinglot=
Language= en
State= Ring (4)
Rings= 1
NativeFormat= (alaw)
WriteFormat= alaw
ReadFormat= alaw
RawWriteFormat= alaw
RawReadFormat= alaw
WriteTranscode= No
ReadTranscode= No
1stFileDescriptor= -1
Framesin= 0
Framesout= 0
TimetoHangup= 0
ElapsedTime= 0h0m0s
BridgeID= (Not bridged)
Context= from-external
Extension= 039988120
Priority= 2
CallGroup=
PickupGroup=
Applica...
2007 Sep 20
0
Video doesn't work for outgoing call?
...l --
Name: SIP/403-097cc8e8
Type: SIP
UniqueID: 1190258125.131
Caller ID: 555
Caller ID Name: (N/A)
DNID Digits: (N/A)
State: Up (6)
Rings: 0
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x2 (gsm)
ReadFormat: 0x100 (g729)
WriteTranscode: Yes
ReadTranscode: Yes
1st File Descriptor: 38
Frames in: 494
Frames out: 510
Time to Hangup: 0
Elapsed Time: 0h0m13s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: macro-play
Extension: s
Priority: 3
Call...
2015 Apr 08
1
Help debugging a possible SIP channel leak in 11.17.0, possible race condition
Have you tried Asterisk 13? The bridging mechanism has been completely
rewritten on Asterisk 12, so there's no longer channel masquerading and
zombie channels. Might be worth a try.
2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>:
> El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?:
>
> I am trying to collect enough information about an