Re: [asterisk-users] Voicemail asking for login

Hi D'Arcy > On 18/04/2017, at 5:17 am, D'Arcy Cain wrote: > > > One user (that we know of so far) has a different experience. In that case > they are asked for a mailbox number first. > > I have tried searching for this issue but nothing seems to apply. Most >

Re: [asterisk-users] Voicemail asking for login

Does he have the same voicemail context?

[asterisk-users] Can't compile Asterisk on Ubuntu 16

All; I am trying to build and install certified Asterisk 13.13 cert3 on a Ubuntu 16.04.2 LTS host without much success. I am getting the following errors when I try to compile. [CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o res_pjsip/config_transport.c: In function

Re: [asterisk-users] PBX selection

On 18 April 2017 at 09:40, J Montoya or A J Stiles wrote: > > It is always preferrable to compile your own Asterisk to fit your hardware and > include just the bits you want, rather than rely on anyone else's pre-compiled > package. Feel free to take a look at

Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16

Feel free to take a look at https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-14-on-Ubuntu.md Ignore the bit about Voipfone and just skip to the "Install Asterisk" bit. I've used this same script with Asterisk 12,13 and 14 on Ubuntu 15,16 and 17 so this

Re: [asterisk-users] PBX selection

On Mon, Apr 17, 2017 at 10:57:27PM +0800, Speed Boy wrote: > Hi all, I'm new to VoIP, now we have a project that needs a > PBX with client APPs. > In our team we have argument for choosing PBX. By so far, we > have following candidates: > > A: Open source > > 1) Asterisk PBX

[asterisk-users] Crashes in jitterbuffer with framedata->timer_interval > 1000

I had three crashes this morning on a divide-by-zero, for example at abstract_jb.c:1008 in 14.3.0. Does this ring any bell to anybody? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the

Re: [asterisk-users] PBX selection

On Monday 17 Apr 2017, Speed Boy wrote: > Hi all, I'm new to VoIP, now we have a project that needs a > PBX with client APPs. > In our team we have argument for choosing PBX. By so far, we > have following candidates: > > A: Open source > > 1) Asterisk PBX (http://www.asterisk.org) (with

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

You need to ensure that traffic to the SIP box is sent to the correct IP. Also if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT and traffic redirection works as is so the Asus router knows it should send the traffic through tunnel and not via WAN. IMPORTANT: Then

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

-- Original Message -- From: "Ernie Dunbar" To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: 19-Apr-17 10:25:59 AM Subject: [asterisk-users] SIP connections over OpenVPN connection get one-way voice. Hi

Re: [asterisk-users] Crashes in jitterbuffer with framedata->timer_interval > 1000

On Tue, Apr 18, 2017 at 1:59 PM, Richard Kenner wrote: > I had three crashes this morning on a divide-by-zero, for example at > abstract_jb.c:1008 in 14.3.0. > > This is quite odd. I took a quick look at the code at that line number and it appears the divider should never be

[asterisk-users] SIP connections over OpenVPN connection get one-way voice.

Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't working *quite* as well as we'd hoped. First, here's our technical details: The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT router. The router has UDP port 1194 forwarded

Re: [asterisk-users] PBX selection

Have a look at xCally from Xenialabs too – they are particularly popular with call centers (and still asterisk based). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roamer2998 Sent: Tuesday, April 18, 2017 11:00 PM To: Asterisk

Re: [asterisk-users] PBX selection

Well said Alex On Tue, Apr 18, 2017 at 7:06 PM Alex Epshteyn wrote: > The solution you choose should be based on many factors which should > include your business requirements, team's experience, your budget, growth > expectations and more. > > You can choose Asterisk or

Re: [asterisk-users] Voicemail asking for login

On 2017-04-18 08:17 PM, Pete Mundy wrote: On 19/04/2017, at 7:58 am, D'Arcy Cain > wrote: Everything looks the same as another one that works except for two things. The one that works doesn't have the "Probation passed" lines. I am not

Re: [asterisk-users] Voicemail asking for login

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

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

On 2017-04-18 03:39 PM, Sebastian Nielsen wrote: You need to ensure that traffic to the SIP box is sent to the correct IP. Also if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT and traffic redirection works as is so the

Re: [asterisk-users] Voicemail asking for login

> On 19/04/2017, at 7:58 am, D'Arcy Cain wrote: > > > Everything looks the same as another one that works except for two things. > The one that works doesn't have the "Probation passed" lines. I am not sure > if that is even part of this call. The other is the line

Re: [asterisk-users] Voicemail asking for login

Hi Darcy, What Pete think is correct. Maybe excecuting the following command at Asterisk console, will help you: asterisk> voicemail show users And you will get a list of all mailbox configured in your system. Search for the user with problems. Finally, in the Asterisk wiki you can find more

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

Sent from my iPhone > On 19/04/2017, at 11:43 AM, Ernie Dunbar wrote: > >> On 2017-04-18 03:38 PM, Duncan Turnbull wrote: >> -- Original Message -- >> From: "Ernie Dunbar" >> To: "'Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] PBX selection

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

On 2017-04-18 03:38 PM, Duncan Turnbull wrote: -- Original Message -- From: "Ernie Dunbar" To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent:

Re: [asterisk-users] PBX selection

Thanks All. Thanks Alex, we also tested thirdlane PBX, and comparing it with PortSIP PBX, Vodia PBX, we hope we can make decision next week. Best regards, On Wed, Apr 19, 2017 at 10:05 AM, Alex Epshteyn wrote: > The solution you choose should be based on many factors which

Re: [asterisk-users] Voicemail asking for login

On 2017-04-18 08:31 PM, Victor Villarreal wrote: Maybe excecuting the following command at Asterisk console, will help you: asterisk> voicemail show users And you will get a list of all mailbox configured in your system. Search for the user with problems. VoiceMail stocktrans2 Angelica