[asterisk-users] sql schema without alembic

hi, is there way to get SQL schema for Asterisk 13.7.0 without alembic? thanks -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] missing https://github.com/asterisk/asterisk/blob/13.7/asterisk-13.7.0-summary

Marek Červenka wrote: hi, there is missing https://github.com/asterisk/asterisk/blob/13.7/asterisk-13.7.0-summary.html is it a mistake or "feature" of security releases summary ? The 13.7 branch is created and used as part of the release process. Tags are made from it. As a result of this it

Re: [asterisk-users] sql schema without alembic

On Thursday 04 Feb 2016, Marek Červenka wrote: > hi, > > is there way to get SQL schema for Asterisk 13.7.0 without alembic? > thanks Assuming you already have Asterisk up and running, you can just use $ mysqldump -d -uroot DATABASE TABLE1 TABLE2 TABLE3 ... will print (on STDOUT, so you can

[asterisk-users] missing https://github.com/asterisk/asterisk/blob/13.7/asterisk-13.7.0-summary

hi, there is missing https://github.com/asterisk/asterisk/blob/13.7/asterisk-13.7.0-summary.html is it a mistake or "feature" of security releases summary ? -- --- Marek Cervenka === --

[asterisk-users] How to simulate 100 phones in a lab ?

Hello, I've got a request from a prospective customer demanding a SIP hardphone able to provide 120 BLF to its operator. Each BLF should show current extension activity (blink when ringing, ...) and allow speed dialing. Beside finding hardware matching these requirements, I'm wondering if it's

Re: [asterisk-users] How to deal with error messages passed as Early Media

2016-02-03 22:12 GMT+01:00 Duncan : > > > On 04/02/16 06:00, Scott Griepentrog wrote: > > For calls that fail, even where early media is played, the call should > terminate with a 4xx or 5xx SIP response which to a certain degree > correlates to the nature of the actual

Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

>>> On Feb 4, 2016, at 12:55 PM, Vitor Mazuco vitor.maz...@gmail.com wrote: >>> so this context parkedcalls is inside on features.conf? Correct. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

Humm thanks very much :) Em 04/02/2016 19:58, "Doug Lytle" escreveu: > > > >>> On Feb 4, 2016, at 12:55 PM, Vitor Mazuco vitor.maz...@gmail.com > wrote: > > >>> so this context parkedcalls is inside on features.conf? > > Correct. > > Doug > > > > -- >

[asterisk-users] NAT traversal for mobile app softphones - best strategy?

Greetings, My asterisk systems sit behind a Meraki mx80 firewall at a data center. I use static public IPs on the firewall and port forward 5060,5061, and 10,000-20,000 so the clients can connect. Per Meraki support: "Our MX security appliances do not support SIP ALG. Our NAT is a

Re: [asterisk-users] What is SIP Early Media useful for ?

I've always wondered if early media with video would allow to see who is calling before answering. Is it possible? Is anybody doing this? On Wed, Feb 3, 2016 at 11:56 PM, Olivier wrote: > Hello, > > Could you help me to summarize what is SIP Early Media useful for ? > > I was

Re: [asterisk-users] What is SIP Early Media useful for ?

I think it's very difficult: most SIP communications are done with SDP data passed in INVITE message. So basically, Callee is able to send Early Media to Caller but the opposite can't happen until Callee has sent back its SDP data, which happens when Callee accepts the incoming call. Maybe a

[asterisk-users] Peer Reachable / Unreachable on TLS

Hello Asterisk users :-) Server: Asterisk 11.7.0~dfsg-1ubuntu1 on Raspberry Client: Zoiper on Android device If "Transport=tcp" everything works fine, without any trouble. If "Transport=tls" registration is fine. But after a few minutes, the peer continuously is "reachable...unreachable".

Re: [asterisk-users] include => parkedcalls but nonexistent context 'parkedcalls'

Humm, so this context parkedcalls is inside on features.conf? 2016-02-03 17:42 GMT-02:00, Doug Lytle : On Feb 3, 2016, at 2:32 PM, Vitor Mazuco vitor.maz...@gmail.com wrote: > Ah no, I'm asking what code I put inside of parkedcalls? > > Nothing, > > The context

Re: [asterisk-users] How to simulate 100 phones in a lab ?

You could use Custom Device States to create your 100 extensions to watch and update their status by changing the states manually. With Asterisk 13.0 and PJSip I had the RLS feature working with some Yealink phones (T28P and a T41P to be specific), on the phone I set:

[asterisk-users] Call hangup on transfer when originated from a Queue

Hi all, i'm writing because going crazy on this issue i'm unable to solve. My VoIP system is based on OpenSIPS router that forward calls to an Asterisk BOX to have IVR and Queue services. If a call was directed to a queue and operator answer, on transfer to another ext. the call hangup. On