similar to: Web-driven SIP call thru Asterisk IPBX

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client

Hi, I am looking for a hosted / virtual IPBX *PLATFORM* for service provider.Such hosted IPBX platform is aimed to be as a service, so that final customers don't have to install, maintain, or upgrade any PBX hardware or software. It should have a control panel for end users to create / edit extension, conference rooms , IVR menus, etc Could you please indicate companies that offer such

I just wondered if there is anything planned along similar lines for Samba4? It would be massive to be able to synchronise users but the problem with password hashes and different algorithms is a real headache. I have only just started with 389 Directory server and wow after OpenLdap its really impressive and such a doddle to use and administer. They have this winsync plugin that opens up an

I've been trying to install R into a user's home directory for them by compiling from source code, on a machine for which neither of us has administrative access. I've run configure using the --prefix option to specify their home directory. This seems to work OK up to a point, but when running make, it seems to expect to be able to install IDL into /usr/local/, and if you are unable

The documentation of how to use the 79xx series' phones and features with Asterisk is really hard to find and put together. The higher end phones like 7970 are more like converged PC+phone, a thin client to telephony and network apps. But it's really hard to target it as a development and deployment platform because the docs and techniques are so obscure. There seems to be a fair amount

Missed this in the [patch] state. If we are gonna do this can we get a timestamp on every open? or every message? -JX On Oct 25, 2006, at 5:30 AM, Xen staging patchbot-unstable wrote: > # HG changeset patch > # User kfraser@localhost.localdomain > # Date 1161768423 -3600 > # Node ID 0c7923eb6b9846c92f1c15486e06ee9745bcf676 > # Parent 410df40afc014555ce7bfcad2852de9e6d0425f0

I have found that specifying different "sizes" in the sample command has a funny effect on the random sampling. The code below is a condensed version of a function I wrote to simulate a bootstrap method. For simplicity, I eliminated the internal bootstrap loop, but kept a statement to draw one bootstrap sample, because this is where the problem occurs. The output (mean(y)^2) should be

Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Anyone know what I need to do to get the FXO port on the SPA 3000 to forward calls to Asterisk? My Asterisk is running on port 5061 and I set the dial plan on the device to forward to s@asteriskip:5061 but Asterisk is not picking it up. I can see on tcpdump traces that the Invite packets do go to through to the asterisk machine on port 5061, but it's not picking them up. sip debug does not

Hi ALL, I'm using this TNT to front-end an asterisk cluster, working pretty well so far. Some T1's are inbound from PSTN PRI's and others are Outbound to PSTN PRI's. Specifying what traffic to send out what PRI is pretty easy, we have unique trunk numbers assigned to specific T1's or groups of T1's, so when I send SIP traffic to the TNT, I prepend the dialed call with

Hi, I'm using asterisk 1.2.1 on a debian sarge distro. I've followed notes in http://www.voip-info.org/wiki/view/Polycom+auto-answer+config and http://www.voip-info.org/wiki/index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO but I still cannot change ring style via asterisk using exten => 666,1,SipAddHeader(ALERT_INFO="ring3") in extensions.conf . Is it

Hi perhaps someone else already had the same problem. Problem description: I''m running a class B University network with approx 10k hosts attached. I would now like to stop traffic from and to hosts in my network not already registered in my DNS server. This means I''ve to handle with approx 50k rules|routes. Sure I can summarize the unalloctaed address space a little bit with

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the

typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. *same => n,Read(mobileNumber,app/input-mobile,10,,2,15)* In the logs: When it fails: - - <SIP/ipbx-iwred-000002e> Playing 'app/input-mobile.slin' (language 'fr') - - User disconnected When it succeeds: - - <SIP/ipbx-iwred-000002e> Playing

Folks! Netweb Group, Inc. fully supports connectivity to any Asterisk PBX systems you have and can provide A-Z termination with immediate effect. Any volume is good enough for us, even an amount as small as $1.00 a day will do for us. We will provide connectivity from our Softswitch IP 216.162.116.46. If anyone is interested, add this to your Asterisk IPBX and then email me for setting up a

Hi. When I call my RNIS numbers (with a mobile phone for example), I can see 2 incoming calls on the IPBX, which should not happend. I'm not sure if it's a problem with the telco France Telecom and their ISDN setup, or if it's a problem with the MISDN driver on the IPBX itself. I'm stuck ... Any advices for troubleshooting that? Someone provide working configuration files

I would like to ask, if someone observe also problem with peer qualify problems, my asterisk log is full with UNREACHABLE/REACHABLE messages, even when two asterisks are in LAN environment, please take a look into this debug, I can't find any problem with packet loss, all qualify requests are replied and acknowledged, I will submit bug report, if you will also not find any problems here...

Hi list, I write on the list looking for help, buy a openvox gw gsm for four channels and I'm a little disappointed with the support openvox, for some reason , The call doesn?t get trough support tells me it was my asterisk server, but does not really work me and my internal calls are working perfectly, I tested with another sangoma FXO gateway and works perfectly. the problem is that

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia