similar to: PlayTones not working

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

On Tue, Jul 4, 2023 at 7:52 PM TTT <lists at telium.io> wrote: > Building on my last message, I am trying to get CHANNEL data using getvar > (through the AMI). And although I'm getting responses, some values > returned seem illogical. For example, phone 111 calls phone 222 via the > PBX. Here's the data I get back > > > > > > Channel A:

Channel A: "1688509741.112" , name: "PJSIP/111-00000064" , is originator: Y , call-Id: "u.l6kcou25cax60 at mydomain.com <mailto:u.l6kcou25cax60 at mydomain.com> " , local_uri: "<sip:222 at mydomain.com <mailto:sip%3A222 at mydomain.com> ;user=phone>" , local_tag: "1734d973-c4da-4ae8-a37d-5f7065f1fe54" , local_addr:

Building on my last message, I am trying to get CHANNEL data using getvar (through the AMI). And although I'm getting responses, some values returned seem illogical. For example, phone 111 calls phone 222 via the PBX. Here's the data I get back Channel A: "1688509741.112" , name: "PJSIP/111-00000064" , is originator: Y , call-Id: "u.l6kcou25cax60 at

Am 15.06.2020 um 21:50 schrieb Luca Bertoncello: > What do you mean now? If I can use the full available band or if I can > download exactly 50Mbs? > The answer to the first question is: YES! That's why I use a traffic > shaper... ;) > The answer to the second question is: NO. I made a speedtest right now > and I get only ~18Mbps download. And some other information, too.

Hello Luca, We are still playing with visualization of your data, but I didn't want you to wait any longer for some results.  I think I blame both DT and the Pi :) First, a look at the phone side of your Banana Pi.  The first thing we noticed is there were a LOT more packets in one direction (north towards DT) than the other (towards the phone): jeff at

Hi List, I?m having problems compiling chan_misdn: asterisk:/usr/src/chan_misdn-beta-0.0.3-rc4 # make install cc -ggdb -Wall -D_GNU_SOURCE -Wno-missing-prototypes -Wno-missing-declarations -fPIC -I/usr/src/asterisk/include -DAST_CONFIG_DIR=\"/etc/asterisk/\" -I/usr/src/mISDNuser/include -I/usr/src/linux-2.6/include -I/usr/src/mISDNuser/i4lnet/ -Wall -c -o chan_misdn.o chan_misdn.c

I found a clue as to why the second leg is not returning a local or remote address: [2023-07-06 11:40:35] WARNING[253072]: pjsip/dialplan_functions.c:903 channel_read_pjsip: No transport information for channel PJSIP/222-0000007d [2023-07-06 11:40:35] WARNING[935126]: func_channel.c:527 func_channel_read: Unknown or unavailable item requested: 'pjsip,local_addr' [2023-07-06 11:40:35]

The following AMI command works well for all of the information I want: action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-00000028 Variable: CHANNEL(pjsip,XXXX) Where XXXX can be one of the many available items. However, when I create a call from A to B, all of the items return properly except: local_addr and remote_addr. More specifically, they return correctly for the A leg (that

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Because it seems my mail from 30th august didn't make it to the list i send it again. If the mail _did_ get to the list and i didn't see it please excuse the duplicate post Below is the mail from the 30th: I have a setup like this: An asterisk-server with SIP-phones on the outside of a NAT. For example: asterisk with local IP-address

The subject says it all. I'm betting there's a way to do it, but so far I haven't found the dialplan runestone via web searching. Thanks. b.

1) I'm working on a quick replacement for DISA, and I ran into the following snag: When I specify "Playtones(dial)" I can only get around 7 seconds of wait time before the dialtone stops, and the context goes to the "h" extension. Is there a way around this fixed timeout? The DigitTimeout setting doesn't seem to have any effect at all on this hangup problem. I

hi, I have some problem with musiconhold or playtones (background,...) in this context someone dial out thru sipura 3000: Executing Dial("Zap/1-1", "SIP/sipura3000/054419949|20|m") in new stack -- Called sipura3000/054419949 -- Started music on hold, class 'default', on Zap/1-1 -- SIP/sipura3000-61fe is ringing -- SIP/sipura3000-61fe answered Zap/1-1

Hello, I would like the dialing party to know what happened to the call, since asterisk doesn't relay a sip error back to the originating sip channel (would be nice, a if (org_channel = sip && dst_channel = sip, relay error to sip client) I want to set up audio feedback on the call status. I've changed the county setting to NL in indications.conf and created this test

Hi everyone, I'm in the very early stages of rolling out an asterisk box at work, and one of the things I'm setting up is a trap for telemarketers >;) What I want to do is have a sipgate number in the UK here which rings for 10 seconds without calling a hard or softphone, then goes to a voicemailbox. The problem I'm having is that Playtones doesn't seem to be sending any

I've researched my brains out on this, and can't find any answer. Is there a way to adjust the level of the tones generated through the Playtones command? I'm thinking that I may have been approaching this incorrectly by targeting indications.conf since the tones are being called via the Playtones application. My sense is that it's not possible due to the lack of response from

I?ve gotten PlayTones to work, however it stops playing the tones as soon as the call is answered. I would like to use PlayTones during the call because I want to have a tone/beep played in the background while call recording is going on. Anyone know a way to get PlayTones to work while call is in progress? Alternatively, does anyone have a suggestion for playing the tone/beep for recorded

Hello Everyone, We have most of the modules commented out. Can someone please let me know which modules needed to be included for Playtones? Kind Regards, Nick. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150509/15ea3418/attachment.html>

symack wrote: > Hello Everyone, > > We have most of the modules commented out. Can someone please let me > know which modules needed to be included for Playtones? The PlayTones application is in the app_playtones module. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org