similar to: MeetMe transcode / format problem

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

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! 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

We have a variable set for each user/peer/whatnot that signals what the outbound caller-id should be sent as with our carrier. When someone dials a followme extension, this does not appear to be carried over for when the calls reach an outside caller, and we see the outbound caller-id being set as 'asterisk' vs the number desired. Has anyone else seen this, or found a way to

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

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

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

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: #

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 list can U help me caller id in USTM if now working -- Starting switch on '4211 at 4211-1' to 4203 -- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0", "") in new stack Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0: ================================================================================ Info: Name=

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! Running Asterisk 20.0.0 on CentOS 7, logging CDRs using cdr_adaptive_odbc to mariadb-server-5.5.68 (via mariadb-connector-odbc-3.1.7-ga-rhel7) Using chan_sip. I'm facing the problem when there is a sudden spike of calls, some of the calls that are being made during those spikes hang forever basically. This looks like this: [root at voip]# asterisk -rx 'core show channels

Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. <--- Received SIP request

I've tried to put a call file to /var/spool/asterisk/outgoing/ to make an outgoing video call, but not succeeded. I could hear the audio, but no video. The asterisk version is 1.4.10, with videosupport=yes The client is eyebeam 1.5.7, with h263 support. Here are some debug messages. It shows the client and asterisk negotiated the video capabilities without problem. However, the 'show

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

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

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

Hello to all. I do not know what is causing choppy music on hold when call comes in through E1 card (PRI).. but this channel info is somehow strange.. We use Alaw over PRI (and I think it's format number 8), But why is WriteFormat at 2 ????? Thanks! show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: 1136667936.0 Caller

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