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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 supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm > > Yes, the GUI is not tha...

Max Grobecker <max.grobecker at ml.grobecker.info> schrieb: Hello Max, > I'm also a customer of the DTAG. > Yesterday, the messed a bit with their DNS entries... Maybe they tried again to repair a working system... :) > If you are NOT using their DNS resolvers you got a "wrong" I...

...g is the call duration reported by asterisk was just few seconds while the call duration reported by the provider was few thousand seconds, the max allowed. So they will be able to terminate the call on the asterisk side and have it run on the provider side. Leandro 2016-09-15 19:18 GMT+02:00 Max Grobecker <max.grobecker at ml.grobecker.info>: > Maybe the client just put the call on hold. > So the call technically has not ended AND the client does not need to send > or handle any RTP data. > Is there any mention of "music on hold" for this channel? > > Greetings &gt...

Thank you very much, Max, for this valuable and informative answer. Offline billing must be quite complex to set up as several telco may be involved (or origination,transit or termination). Moving to normal landline fare seems much simpler ! Thanks again 2017-03-28 21:41 GMT+02:00 Max Grobecker <max.grobecker at ml.grobecker.info>: > Hi, > > in Germany, this kind of regulation is in effect for phone numbers which > cost more than a normal landline call. > The regulation states, that the waiting time must not be charged to the > customer. > > > Most compan...

...the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker

Hi all I'd like to log the client IP addr and port used for SIP and RTP *during* in a call. The IPs must be the real source IPs (internet accessible). How are these parameters available from dialplan? For instance, ${SIPURI} holds the internal "IP:port" if the client is behind NAT. I need the external IP:port Regards Ethy

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:

Related to http://lists.digium.com/pipermail/asterisk-users/2016-November/290384.html, at the moment I'm passing one variable via DIAL. Now I'd like to pass a whole bunch, and my idea was to rather than having a great string of b(synctest3b^setVar^1(something)^2(more things)^3(etc)) and then get them with ARG1..ARGn etc, I could bundle the whole lot into a HASH and then unbundle them at

Hello, I've got a problem at the moment, that setting "transmit_silence = yes" seems to have no effect on Asterisk 1.8-Certified. Although it's enabled and "core show settings" confirms, that it is really enabled, there are no RTP packets sent by Asterisk when waiting for DMTF input or when "Wait()" is called. Also, there seems to be a small gap of 2 or 3

Hi list! Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't connect to the remote Server (by Telekom) until today about 7:30. Well, it could happen... What I find really annoying was that I needed to restart Asterisk as I checked with sipsak that the Telekom-Server works... I think, this should not be normal... Can someone explain me why it happens and what I have to

https://bugzilla.netfilter.org/cgi-bin/bugzilla/show_bug.cgi?id=106 laforge@netfilter.org changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |ASSIGNED ------- Additional Comments From laforge@netfilter.org 2003-06-30 17:12 ------- can you please try to use

I am banging my head over a simple asterisk trick I was seeing on one asterisk server. An extension dials an international premium number, the called number answers, then the extension hangups, but the call continue to run on the international number side, generating an high profit for the premium number company and a big loss for the asterisk owner. I think some sort of "transfer"

Hi, I recently installed some fresh CentOS 6.5 machines and it took only about 20 minutes until the file system (ext4) was broken. And with "broken" I mean, that the system wasn't able to find vital system libraries any more! I were able to reproduce it on highly different systems: - A fresh installed CentOS 6.5 64 Bit on a virtual machine (KVM) - A system which I installed some

...***In channel:PJSIP/6001-00000009 sharedVar: Answered channel:PJSIP/6001-0000000a ***In channel:Local/s at svtest2-00000032;2 sharedVar: "I have been set in svtest2" ***In channel:PJSIP/6001-0000000a sharedVar: "I have been set in svtest2" On 27 November 2016 at 11:39, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hi, > > is channel variable inheritance working for your setup? > Passing variables to other channels can normally simply be done by naming the variable with one or two prefixed undersorces > to make it available to the channel that...

Hello, I'm running a Squid cache (Version 3.1.10) on CentOS 6.6 as a forward proxy which is reachable over a global IPv6 address. For whatever reason, Squid tries to perform PTR lookups on the client's IPv6 address. The weird thing is, that Squid seems to struggle with the "endianess" of the IPv6 address blocks. For example: My current client IP is

Hi, I'm trying to use AGC in combination with Asterisk 1.8 and an odd telephone which is very loud when used with a headset and more quiet when used "normal". Regarding to the documentation, AGC should be available since * 1.6 - but every time I want to set it, the CLI tells me: -- Executing [0160xxxxxxx at intern:2] Set("SIP/intern-xxx-000000d2",

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

On 2016-11-22 07:49 PM, Pete Mundy wrote: > > One direction that may be worth exploring further is his ATA's config (or perhaps swapping it for a different model). Eg adjusting echo cancellation or line impedance settings. I have to be careful here as I auto-provison these devices and changes would propogate to every user. Echo cancellation is off. Do you think it should be on?

Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 -> IVR answers and puts caller to the chosen queue -> Someone picks up the phone (Internal ext. 321) -> CallerID shown on customers

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162