similar to: asterisk too many files or memory leak???

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

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

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

> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.

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 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.

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 am hoping someone else has seen this and can offer a solution or at least a direction to investigate. I am running 11.23. Most of my clients are fine but one has a strange behaviour. He has a Grandstream HT701 like most of my clients who use an ATA. He can make call and they are crystal clear. However, when he tries to use phone menus ("dial 234 for John Doe" for example) it

Hello, I am using Asterisk voicemail on a CentOS 7 server. I would like to be able to remove the 'mailbox' prompt and 'password' prompt when a user tries to access their voicemail. I can remove the 'password' prompt by not setting a password for the user, but the 'mailbox' prompt is always heard. Please let me know how Asterisk can be configured to remove these

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 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>

My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. What I want to do is execute a remote command with the voicemail as an argument. The remote machine command would email the message. I'm thinking of: same =>n,VoiceMail(vm,u) same =>n,System(ssh myserver "emailVM

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

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

I'm debating between a cloud PBX or, perhaps, rasberry pi. For a SOHO, maybe three hardphones, rasberry pi would suffice? I would be amazed, but, if so, great. thanks, Thufir -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160706/84a0ac7c/attachment.html>

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. > >

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > Make sure you have solved the problem. You don't want to get hit with a > phone bill for calls from your location to Israel. Basically, they are > hoping that you are running the equivalent of a mail server open relay. > They are trying to use you to dial out to another number. You don't want > to pay

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 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 =>

>>>>> D'Arcy Cain <darcy at VybeNetworks.com> writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos <cloos at jhcloos.com>