search for: haeber

Hi Barry Flanagan, Barry Flanagan <barryf-lists at 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

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

Hi Stefan, Stefan Tichy <asterisk3 at 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 sure? Yes, im sure. I have looked at the time and talked for at least 4 seconds. In CLI log are 5-6 seconds visible between...

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 [s at macro-voicemail:77]VoiceMail("SIP/XY-0005175a", "aNumber,su") in new stack [Jan 26 15:24:04] --

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 sure? The value for silencethreshold (140) is unusually large. -- Stefan Tichy ( asterisk3 at pi4tel dot de )

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 | \ | \ Proxy1 Proxy2 I have tries to solve this problem with two trunks for this proxies and Dial(... at proxytrunk) but on this way the

Steve, would you be willing to share that "quick bash script"? James Cass <http://goog_987864563> jcass78 at gmail.com On Wed, Aug 19, 2015 at 12:11 PM, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Wed, 19 Aug 2015, Dominique Haeber wrote: > > Hi Barry Flanagan, >> >> Barry Flanagan <barryf-lists at flanagan.ie> schrieb am Mit, 19. Aug 11:06: >> >>> SIPP is probably what you seek. http://sipp.sourceforge.net/ >>> >>> Hope this helps. >>> >> >> That l...

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-00000086]: sip/sdp_crypto.c:285 sdp_crypto_process: Accepting crypto tag 1 [May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:310 sdp_crypto_offer: Crypto line: a=crypto:1

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&SIP/device2,20) Level3: Dial(SIP/device1&SIP/device2&SIP/device3,20) When in level one, no one accept the call until the timeout, they have a missed call on device. When in level two,

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 at macro-redirection:85] Dial("SIP/access-trunk-00000001", "Local/0XXXXXXX40 at x-xxx-companyXXXXX/n") in new stack [...] [Apr 1