Re: [asterisk-users] asterisk 13 n-way call problem

I spent some time reading docs and such change is not documented, so this is bug. I'll open issue... 22.12.2015 10:53, Dmitry Melekhov пишет: Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in a

[asterisk-users] asterisk 13 n-way call problem

Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in asterisk 11: -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer) priority 1 -- Executing [0@fromtransfer:1] NoOp("OOH323/

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

"Brian ::" schrieb: > sip trace? Could you please explain? I'm not a VoIP-expert... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Jo

[asterisk-users] map softphone ids to asterisk ??

In setting up the GS-Wave softphone there are two id entries: SIP User ID SIP Authentication ID I would have thought SIP User ID was the devicename , i.e. [name]. Then SIP Authentication ID was defaultuser. But not so. With [gs_5062](cell-phones) defaultuser=gs_62 and SIP User ID

Re: [asterisk-users] spa112 can't get line 2 to register

I've got SPA112's sitting behind NAT's for fax use. I left the SPA's on the default config on the latest firmware (1.4.0), then just used the quick setup to configure it. For the extension definition I just specified rport,comedia (or nat=yes on older setups). Works great. -Original Message--

[asterisk-users] spa112 can't get line 2 to register

Hi all, I've got a couple of SPA112's that are having problems registering line 2. Line 1 registers just fine. All of them are behind a NAT. Here is a sample provisioning file that the devices are using. (Any help would be most appreciated.) Yes xxx syslog.example.com 3 Yes Yes 2 360

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

sip trace? On Mon, Dec 21, 2015 at 6:56 PM, Luca Bertoncello wrote: > Karsten Wemheuer schrieb: > > Hi Karsten! > > > the timeout value of 15 minutes directs me to an issue with session > > timer. Try to refuse them by putting the line > > session-timers = refuse > > into the general co

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

Karsten Wemheuer schrieb: Hi Karsten! > the timeout value of 15 minutes directs me to an issue with session > timer. Try to refuse them by putting the line > session-timers = refuse > into the general context of sip.conf. Reload the sip stack with "sip > reload". Sorry, I forgot to ment

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

Hi Luca, Am Montag, den 21.12.2015, 18:52 +0100 schrieb Luca Bertoncello: > Hi list! > > My Problem: all calls to international numbers will be dropped after exactly > 15 minutes... > I have a VoIP-account by Deutsche Telekom. > This is what I see when I call someone (my parents) and the connect

[asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

Hi list! My Problem: all calls to international numbers will be dropped after exactly 15 minutes... I have a VoIP-account by Deutsche Telekom. This is what I see when I call someone (my parents) and the connection will be dropped: == Using SIP RTP CoS mark 5 -- Executing [+3901522@defau

Re: [asterisk-users] weather.agi

Jeff LaCoursiere writes: > On 12/16/2015 11:24 AM, Ryan Crowder wrote: >> http://www.wunderground.com/weather/api/ > > Awesome! Perfect! > > Cheers, > > j > >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Be