search for: writetranscode

...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: ${confnum}\n" > /var/spool/asterisk/outgoing/${number...

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...

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) ???? Read...

...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 Picku...

...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) Varia...

...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...

...-- 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...

....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 Pick...

...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 Gr...

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

...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 G71...

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

.../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=...

...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 PS...

...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 --...

...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/+49xxxxxx...

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

...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= Applicat...

...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 G...

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