similar to: Cisco SX20 disconnecting before call

Morning, We recently pushed our Asterisk video bridge into a DMZ and since then, local calls have been unreliable to say the least. While offsite calls work nicely, calls on our internal server usually fail to ring the far end. Two test calls that were made 4 minutes apart yielded different results: one rang the far end, the other kept trying to transmit the Invite. The configuration didn't

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, 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 testing I found several bugs

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1&t=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching 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

Hi all, Cisco adopts AAC-LC in its TelePresence product, why we can't use AAC-LC instead of celt? Steven -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20101014/094f4ef9/attachment-0002.htm

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

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?

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's say 200 then he will

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Morning! I'm trying to subscribe a softphone to an Asterisk 11 server, but it sends an "ua-profile" event that Asterisk immediately rejects with a 489 Bad Event error. Is this event not supported at all? Are there any workarounds? Best regards, Norman -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Jul 2, 2007, at 9:48 PM, Jean-Marc Valin wrote: > Selon zmorris@mac.com: >> But if the echo canceler IS using each frame's timestamp when it's >> trying to converge, it's almost guaranteed to fail on most operating >> systems, because the timestamp has such a high variability between >> frames, and can even sometimes be 0 for the output buffer in this

Hi I was trying to have realtime voicemail working with ODBC Driver. Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page ) But I do not manage to make it work with ODBC. Outside Asterisk, ODBC works fine, I can access my databases & tables ! Asterisk fails to

Hello, I'd like share with you my tests about PJSIP channel with the aim of improving the functioning of the channel: * Multi domain support not 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

Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. On 21-04-17 12:28, Marcelo Terres wrote: > Did you try to activate DEBUG and set the verbosity to a higher level > (100?) to check what Asterisk tells you about? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at

Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to 1111 when I go

Hi, I'm trying to get OpenSSH to work on Solaris 10 wich Sun C 5.8 compiler (SUNWspro 11). I've compiled OpenSSL 0.9.8a without problem and OpenSSH 4.3p2 as well. [user at compilationserver ~/openssh-4.3p2] ./ssh -V OpenSSH_4.3p2, OpenSSL 0.9.8a 11 Oct 2005 My problem is that I cannot connect to anything. When I try I always get an error [user at compilationserver ~/openssh-4.3p2]

Hello My friends, Today my asterisk stop working and i could see the following messags in /var/log/asterisk/messages at the time that asterisk stop working: [Feb 16 13:23:40] NOTICE[8230] chan_sip.c: Peer '324' is now Reachable. (2ms / 2000ms) [Feb 16 13:24:41] NOTICE[8230] chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds [Feb 16 13:25:54]

Hello I have a large cluster in which every node is logging: I [socket.c:2474:socket_event_handler] 0-transport: EPOLLERR - disconnecting now At a rate of of around 4 or 5 per second per node, which is adding up to a lot of messages. This seems to happen while my cluster is idle. -------------- next part -------------- An HTML attachment was scrubbed... URL:

This ''MSN'' is a web site? Im guessing it ''refresh''es every 5 minutes or so. They are proably testing cookies against the ip address they appear to be comming from. This is horribly wreckless of them if they arnt offering IPv6. Are they? They only way i have to remedy this problem is to get their IP range and bind it to the most stable connexion you have,

Hi, just "cvs update"'d to get the latest portable version, to start rebuilding our AIX systems to get zlib-1.1.4 and the channel-bug fix. SSH protocol 2 seems to work nicely, ssh protocol 1 doesn't work properly. Environment: AIX 4.3.3, openssl 0.9.6c, openssh as of today (Mar 12, 11:20 GMT). Client/blowfish, to openssh 3.0p1 or to 2.5.1p1: debug1: Encryption type: blowfish