[asterisk-users] Dial()-Function

Hi all! :) I search a function or option for application Dail(). My situations: I have two or more Dial()s with multiple devices (Handgroups). Level1: Dial(SIP/device1,20) Level2: Dial(SIP/device1/device2,20) Level3: Dial(SIP/device1/device2/device3,20) When in level one, no one accept the

Re: [asterisk-users] Dial()-Function

hi jg, jg schrieb am Don, 17. Mär 14:05: > Wouldn't it be easier to use a local channel and do something like > is done in the "Delay Dialing Devices Example"? > > https://wiki.asterisk.org/wiki/display/AST/Delay+Dialing+Devices+Example No, i think unfortunately

Re: [asterisk-users] asterisk server stress test

Hi Barry Flanagan, Dominique Haeber dominique.hae...@xig.ch schrieb am Mit, 19. Aug 15:13: Barry Flanagan barryf-li...@flanagan.ie schrieb am Mit, 19. Aug 11:06: SIPP is probably what you seek. http://sipp.sourceforge.net/ Hope this helps. That looks pretty like what I'm looking

Re: [asterisk-users] asterisk server stress test

Hi Barry Flanagan, Barry Flanagan barryf-li...@flanagan.ie schrieb am Mit, 19. Aug 11:06: SIPP is probably what you seek. http://sipp.sourceforge.net/ Hope this helps. That looks pretty like what I'm looking for! Many thanks! Sincerely, Dominique Haeber

[asterisk-users] asterisk server stress test

Hi all, i need to test how many calls can withstand an Asterisk server. Do you know any good tools to strain the server? At best, there are scripts that I can run on a Linux server. I thank you for your tips Sincerely Dominique Haeber

[asterisk-users] ReceiveFax() fails over Dial()

Hi all, since asterisk 11 (1.6 was okay) failed the ReceiveFax-Application when it called about Dial and a Local-Channel. Directly from external to FaxReceive is no problem. Cut from cli: [...] [Apr 1 11:12:31] -- Executing [s@macro-redirection:85] Dial(SIP/access-trunk-0001,

Re: [asterisk-users] asterisk 11.14 - voicemail incorrect duration

Hi Stefan, Dominique Haeber dominique.hae...@xig.ch schrieb am Die, 27. Jan 08:55: I have looked at the time and talked for at least 4 seconds. In CLI log are 5-6 seconds visible between open to writing and Hang up. Nevertheless, Asterisk writes about two seconds. The value

[asterisk-users] asterisk 11.14 - voicemail incorrect duration

Hi all, i use asterisk 11.14.0 and I suspect that the voicemail application counts the time wrong. In my voicemail.conf: [general] minsecs=3 maxsilence=5 format=wav maxsecs=180 silencethreshold=140 [...cut..] In the asterisk-cli: [Jan 26 15:23:49] -- Executing

Re: [asterisk-users] asterisk 11.14 - voicemail incorrect duration

Hi Stefan, Stefan Tichy asteri...@pi4tel.de schrieb am Mon, 26. Jan 23:56: Hi Dominique On Mon, Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote: So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only count 2. What can be the reason? It is not silence. Are you

Re: [asterisk-users] deactivate SRTP in asterisk 11

On Fri, 09 May 2014 17:37:14 +0200 jg webaccounts...@jgoettgens.de wrote: Either you do not compile the srtp module into the Asterisk package or you disable RTP encryption on a phone by phone basis. Thank you for your help :) jg dominique --

[asterisk-users] deactivate SRTP in asterisk 11

Hi all, i try to deactivate SRTP in asterisk 11. In sip.conf: tlsenable=no encryption=no transport=udp srtpcapable=no but when I try to make a call comes following message: [May 9 15:19:03] DEBUG[24745][C-0086]: sip/sdp_crypto.c:285 sdp_crypto_process: Accepting crypto tag 1 [May 9

[asterisk-users] Use SRV for failover proxy

Hi all, is it possible that asterisk uses two proxies with SRV? The enddevices are registered on one of the two Proxies (Kamailio). The two proxies communicate with each other. And asterisk can choose one of this proxies with SRV. asterisk | \ |\ Proxy1Proxy2 I have tries to

Re: [asterisk-users] Use SRV for failover proxy

Gareth Blades mailinglist+aster...@dns99.co.uk schrieb am Fre, 06. Sep 13:21: No asterisk will always use the first SRV record and wont load balance or switch to a backup if its not reachable. hmm okay :O What we do is have each endpoint defined in sip.conf with qualify=yes and then in the