similar to: One way audio - doesn't seem to be NAT issue

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

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

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 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.

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

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>

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

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

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. > >

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

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

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 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 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 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote: > This is just screaming "configuration mismatch" -- or, possibly, "latent bug > whereby things parsed in separate places should be treated the same, but are > actually getting treated differently". I really don't want to be the "my system isn't working so there must be a bug in Asterisk" guy

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>

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