search for: nativeformats

...the Sipura from Asterisk). >>>>> >>>>>This patch should be applied to indications.c under the main asterisk >>>>>source directory (usually /usr/src/asterisk): >>>>> >>>>>68a69 >>>>> > if (!(chan->nativeformats & AST_FORMAT_SLINEAR)) return 0; >>>>> >>>>>Oh, and finally, here's a shameless plug to a good friend's website >>>>>(which includes a VOIP forum!): http://outcast.ws >>>>> >>>>>Comments? Questions? :) >&gt...

...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 ------------------------------------------------------------------------ ------------------------------------------------------------------------ -------------- next part --...

...all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this 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: Confb...

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 WriteTranscod...

...TMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- 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 --...

...ast stream in progress, I am typing 'core show channel 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: slinReadForma...

Hi all, if i enter a "show codecs" at cli * response with: 1 (1 << 0) G.723.1 2 (1 << 1) GSM 4 (1 << 2) G.711 u-law 8 (1 << 3) G.711 A-law 16 (1 << 4) MPEG-2 layer 3 32 (1 << 5) ADPCM 64 (1 << 6) 16 bit Signed Linear PCM 128 (1 << 7) LPC10

...UniqueID: 1398809161.20186 LinkedID: 1398809161.20186 Caller 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 Ap...

...actual i have a transcode problem and i have no idea to solve this. All my 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 M...

Hi In asterisk channel ,I so number of variable regarding the Codec ,Can anyone explain what are those variable variable means.Below are the variables 1. chan->readformat 2. chan->writeformat 3. chan ->rawreadformat 4. chan ->rawwriteformat 5. chan->nativeformats Thanks Nikhil

...ch agent is logged on: # asterisk -rx "show channel SIP/303-b2f1c368" -- 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 --...

...XXXX,105,Playback(tt-monkeysintro) exten => _9XXXX,106,Hangup my chan_sip.c: static int sip_write(struct ast_channel *ast, struct ast_frame *frame) { struct sip_pvt *p = ast->pvt->pvt; int res = 0; if (frame->frametype == AST_FRAME_VOICE) { if (!(frame->subclass & ast->nativeformats)) { --> --> ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %d (read/write = %d/%d)\n", frame->subclass, ast->nativeformats, ast->readformat, ast- >writeformat); return -1; } Related error reports I found: http://www.mail-archi...

....226 6 Y 7910 1 The problem is that the phone resets when I attempt to make a call from it or place a call to it. If I pick up I have no dial tone and after 3-4 seconds the phone resets. When that happens, on Asterisk I see: Attempting to Clear display on Skinny 500@test7 skinny_new: tmp->nativeformats=4 fmt=4 -- Starting simple switch on '500@test7' then the phone resets. when I try to call it, it doesn't ring and Asterisk displays: Found device: test7 -- skinny_request(500@test7) -- Skinny cw: 0, dnd: 0, so: 0, sno: 0 skinny_new: tmp->nativeformats=4 fmt=4 -- skinny_call(Skinn...

hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan->nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta

...hese. 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) ???? ReadFormat: 0x4 (ulaw) ?WriteTranscode: No ? ReadTranscode: No -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080808/651777c6/attachment.htm

...law) RawReadFormat= 0x4 (ulaw) or: Name: SIP/pstnlink-000000d1 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 t...

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

Hi, Yet another question, now that I have callerid working correctly, I'm trying to work out how to utilise the different numbers I have. I have a 100 number range allocated to my E1/PRI/OnRamp service. My incoming calls are handled like this: Advertised/published number is an analogue line terminating on a X101P. If the analog line is busy, it has a call diversion to the PRI on a TE405P

Hello, Anyone else experiencing problems with the latest spandsp (pre3) and last libtiff beta? I'm getting 8 bytes long file, with the TIFF header only during such connection: -- Accepting call from 'XXXXXXX' to 'YYYYYY' on channel 0/2, span 1 -- Executing SetVar("Zap/2-1", "FAXFILE=/tmp/foch.tif") in new stack -- Executing