similar to: Touch tone stutter

On 2016-11-22 07:49 PM, Pete Mundy wrote: > > One direction that may be worth exploring further is his ATA's config (or perhaps swapping it for a different model). Eg adjusting echo cancellation or line impedance settings. I have to be careful here as I auto-provison these devices and changes would propogate to every user. Echo cancellation is off. Do you think it should be on?

On Thu, 4 Aug 2016 14:03:39 +0100 Nabeel <nabeelshikder at gmail.com> wrote: > I should add, a password is *always* asked if a password has been set. > There isn't a way to bypass that. Then something is wrong. http://darcy.vex.net/star98.mp3 -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net VoIP: sip:darcy at Vex.Net

This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. # asterisk -x "core show channels" Channel Location State Application(Data) SIP/thinktel-0000000 (None) Up AppDial((Outgoing Line)) SIP/4164251212-00000 4165555555 at LocalSets Up Dial(SIP/thinktel/4165559999) 2 active

I know that it runs on other systems but do other ports get the same attention? I have been running it on a NetBSD server for about a year now and while it mostly works it just crashes from time to time with no explanation or core dump. I have improved the situation by expanding my intrusion detection but it still stops every few days or so. I have a cron job that tests for it and restarts it

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

I have this in my sip.conf: exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}") same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s) same => n,Hangup However, my extensions are set up so that they always show the external number, not the extension: [foobar2](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx callerid=Candace <5555551212>

This is driving me to distraction. I have a switch with multiple clients who are all working fine except for one and I can't figure out what makes them different. I have tried every NAT setting in the ATA (SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different sip ports, different RTP ports and it still fails. I have left the location with it working only to have it fail

I have an extension that looks like this: exten => 5555551111,1,Verbose(Door buzzer calling) same => n,Dial(SIP/user1&SIP/user2&SIP/user3) The idea is that any of the three users can answer the phone to let someone in. The problem is that if, say, user2 unplugs his phone then the call immediately goes to his voice mail and the other two do not have the ability to open the door.

If I remove the password, how can anyone access the mailbox if the 'mailbox' prompt is not played? Nabeel On 30 Jul 2016 3:19 p.m., "D'Arcy J.M. Cain" <darcy at vex.net> wrote: > On Sat, 30 Jul 2016 06:43:47 +0100 > Nabeel <nabeelshikder at gmail.com> wrote: > > I am using Asterisk voicemail on a CentOS 7 server. I would like to > > be able to

I have been banging my head against the wall for weeks now on this one. I have a switch running NetBSD and Asterisk 11.19.0 although I have had this problem on older versions as well. I, and my users, can call out, we can receive calls, quality is excellent but I cannot talk with one user. The different elements are as follows: The switch as described above which is in a server room on the

On Thu, 1 Sep 2016 11:02:57 +0200 Administrator TOOTAI <admin at tootai.net> wrote: > > [Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application > > 'ExecIf' for extension (unauthenticated, 5555551111, 3) > > > > Is there a module that I need to load? > > > > In case it matters I am running Asterisk 11.23.0 on NetBSD 7.0. > >

On 4 August 2016 at 13:18, D'Arcy J.M. Cain <darcy at vex.net> wrote: > > Let's get this straight. You call yourself from any phone in the world > and press '*' while listening to the message, you wind up in your own > mailbox and you believe that means that you don't need a password? Do > you think that the phone system somehow knows that it is you

I have this in my config: exten => _800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/1${EXTEN}) exten => _1800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/${EXTEN}) exten => _NXXNXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/trunk/1${EXTEN}) exten =>

On Tue, 30 Aug 2016 17:56:35 +0200 Administrator TOOTAI <admin at tootai.net> wrote: > Something like > > exten => 5555551111,1,Verbose(Door buzzer calling) > same => n,Set(toRing=) > same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN > USE"]?Set(toRing=${toRing}&SIP/user1) Failed. I checked the online docs and the syntax seems to

Set(CALLERID(number)=XXXXXXXXXX) works here. Also check with your VoIP provider what format they want for the number. (I believe) most accept a 10-digit number, but I seem to remember reading about the odd provider that wanted a leading "1". On Thu, Jun 18, 2015 at 11:47 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Thu, 18 Jun 2015 13:45:10 EDT > kenner at

It's very simple but it doesn't work. Here's the entire script. #! /usr/bin/python import sys env = {} def comm(cmd): sys.stdout.write(cmd.strip() + '\n') sys.stdout.flush() return sys.stdin.readline().strip() while 1: line = sys.stdin.readline().strip() if line == '': break key,data = line.split(':') if key[:4] == 'agi_':

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

On 30 July 2016 at 19:32, D'Arcy J.M. Cain <darcy at vex.net> wrote: > > > Not playing the prompt changes nothing. If someone presses '*' while > listening to your answer message then they are in your mailbox. You > better have a password that they need to enter to continue. I have now tested the 'Unavailable' message by pressing "*" while

On Sat, 15 Aug 2015 16:30:39 +0800 Michael Dupree <michael at easybitllc.com> wrote: > Not 100% ure, but maybe play with the canreinvite or directmedia > settings. Yes! That was it. Just for future searches here is what I did. I added "directmedia = no" in sip.conf. This fixed the issue. I believe that Asterisk was getting confused when one leg was inside NAT and the

I have an error sending to a specific fax number. It may be more than one but this is the one I investigated. It seems the delay for the SIP negotiation in T.38 was initiated after 6 seconds, however, our system sent the BYE after only 4 seconds, possibly cutting the call before all the communication necessary for the negotiation was completed. Here is the trace from our provider showing their