search for: testphones

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.

...ed IP-addresses of course): [general] port = 5060 bindaddr = 0.0.0.0 context = others ;nat specific stuff below externip = 2.3.4.5 canreinvite = no nat = yes qualify = yes ;qos stuff tos_sip = cs3 tos_audio = ef ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;template for voip-testing [testphones](!) context = voiptest type = friend host = dynamic nat = yes disallow = all ;disable all codecs except the ones defined below allow = ulaw allow = g723.1 allow = g729 ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; [2000](testphones) secret = foobar [2001](testphones) secret = foobar [2...

I?m trying to use Playtones to have a tone played periodically throughout phone calls. Unfortunately, I can?t seem to get PlayTones to work. I never hear the audio tones. Here is the output on the Asterisk console. -- Executing [19525553312 at proxy-dial:2] PlayTones("SIP/testphone-00000032", "1400/500,2000/5000") in new stack [2014-10-30 14:28:31] WARNING[23154]:

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

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

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 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