similar to: Bugfix for CVS-HEAD-06/26/04-21:56:45

Hi 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

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

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF 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

Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast 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

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

After an upgrade to Asterisk 12, I'm "collecting" channels. When I enter and then exit a conference room, I see: -- <CBAnn/207-0000067f;1> Playing 'confbridge-leave.slin' (language 'en') -- Channel CBAnn/207-0000067f;2 joined 'softmix' base-bridge <5edb1920-3774-4ba3-8c4d-23e8fd04519c> -- Channel CBAnn/207-0000067f;2 left

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

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

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

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

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

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars =

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

Hi, I am trying to get a prototype working based on Asterisk and Quicknet cards. I currently have to systems set up each with a LineJack card. I have the systems working, but can't get the voicemail demo to work properly. Messages to the user telephone set from the voicemail system are clear, but recordings left through the phone are distorted. Sounds like the audio has lots of echo and

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

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

Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my

I have all three running on the same box. I say OT because it appears asterisk is doing it's job just fine. It must be an IAXmodem or faxgetty (hylafax) problem When faxes work, they look great. I have ten IAXmodems setup with different ports and they register fine. I have ten faxgettys that startup fine. I start the IAXmodems and then faxgettys in inittab. They are setup as a roll