search for: microsip

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...

...ching and late nights since, but have had to admit defeat. Rather than duplicate it all here, I've posted my logs and conf files on that thread, too. Problem is that while there are quite a few sip examples, I have chosen to take the path of pjsip. Seems I can manage to attach Blink, Zoiper, Microsip and my ITSP with multiple extensions without problem to my Asterisk, but sip2sip has beaten me! It's presumably something ridiculously simple, but there comes a point where you can't see the wood for the trees. If someone can help me resolve this, I'll post a complete guide on Github...

Good day, I'm trying to set up a video conferencing solution for our office. Currently, my testing setup has my laptop running MicroSIP and a Debian VM with Asterisk installed on it, with the other end being a Cisco Telepresence SX20 box. Video calls from MicroSIP to SX20 work perfectly, but if I try to place a call from the SX20, it pops up a "Connecting" screen for a second, then the call ends with "Call has discon...

...: 1.1.1.1:62850 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 660081914 SIP Options : (none) Codecs : 0x20010a (gsm|alaw|g729|h264) Codec Order : (alaw:20,g729:20,gsm:20) Auto-Framing : No Status : OK (16 ms) Useragent : MicroSIP/3.15.1 Reg. Contact : sip:660081914 at 192.168.1.104:62850;ob Qualify Freq : 120000 ms Sess-Timers : Refuse Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No sip*CLI> sip...

Hi, Am 16.02.2017 um 14:19 schrieb Annus Fictus: > And Microsip using PJSIP SIP stack :) Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not sure if this really is a sign of good quality. Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP in other software), but after just five minutes of test...

Hello, can someone recommend a good and free Softphone for Windows which does not display advertisments inside the program? We have X-Lite but free version display advertisments. thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140608/c3a9e887/attachment.html>

Well, what immediately stands out is: "FILE * open failed!" Have you triple checked that the full filepath is correct and that the user that Asterisk is running as has full permissions to access your valid certificate file? I have it working with microsip and a free TLS cert from LetsEncrypt. When I get to the PC with that on, I can write up what settings I've got if that helps? On 26 August 2016 at 10:47, hw <hw at gc-24.de> wrote: > hw schrieb: >> >> >> Hi, >> >> I?m trying to get TLS to work with aster...

...t; Allowed.Trsp : UDP >> Def. Username: 660081914 >> SIP Options : (none) >> Codecs : 0x20010a (gsm|alaw|g729|h264) >> Codec Order : (alaw:20,g729:20,gsm:20) >> Auto-Framing : No >> Status : OK (16 ms) >> Useragent : MicroSIP/3.15.1 >> Reg. Contact : sip:660081914 at 192.168.1.104:62850;ob >> Qualify Freq : 120000 ms >> Sess-Timers : Refuse >> Sess-Refresh : uas >> Sess-Expires : 1800 secs >> Min-Sess : 90 secs >> RTP Engine : asterisk >> Par...

Hi first post, so hope I'm not violating protocol. Been using Asterisk as home phone and hobby use for nearly 10 years. I love this project. Anyway, would someone mind verifying my pjsip.conf ? This seems to work well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade to 14.4.1. Other than that the phone registers properly on 14.4.1. I can provide a pjsip log as well,

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

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, I?m trying to get TLS to work with asterisk and client phones, and all I?m getting from asterisk is [Aug 23 11:46:42] WARNING[1170]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 23 11:46:44] WARNING[1171]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! when clients try to

Hi I want IM messages between Softphone clients like Ekiga and MicroSIP. I don't care about SMS/GTALK etc. I only need this. I followed this guys tutorial here: http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html But this wasn't intended for IM messages. It works kinda interestingly. When you send a message to your contact let&...

...t work correctly: https://issues.asterisk.org/jira/browse/ASTERISK-26026 * Different context subscribe for each endpoint not possible: https://issues.asterisk.org/jira/browse/ASTERISK-25471 * BLF don't work correctly on my tests with X-Lite, BRIA, JiTSI. Only work partially with microsip but because this softphone use the same SIP STACK (PJSIP). I test BLF with the latest Asterisk version and latest FreePBX version. The problem is when a softphone is on the phone on the other softphone appear off-line. On Jitsi not is possible know if a Endpoints is or not online...

...0081914 >>>> SIP Options : (none) >>>> Codecs : 0x20010a (gsm|alaw|g729|h264) >>>> Codec Order : (alaw:20,g729:20,gsm:20) >>>> Auto-Framing : No >>>> Status : OK (16 ms) >>>> Useragent : MicroSIP/3.15.1 >>>> Reg. Contact : sip:660081914 at 192.168.1.104:62850;ob >>>> Qualify Freq : 120000 ms >>>> Sess-Timers : Refuse >>>> Sess-Refresh : uas >>>> Sess-Expires : 1800 secs >>>> Min-Sess : 90 se...