similar to: "transmit_silence" not properly recognized on 1.8 ?

Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted programmer_ted@hotmail.com P.S. Read to the very end. The original bugfix

Hello list, I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use. I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP

Hi there I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can't nail down the reason. I don't believe this is a general issue, rather some

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

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

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" IP address back > that was not working. Besides that, you should disable SRV lookups

No, there is no Music On Hold starting and the bad thing 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

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:

Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly

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

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

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",

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, 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

hello, since while recoding asterisk is not sending an audio stream, the remote party times-out rtp and is hanging up on us. is it possible to send a blank audio stream while recording app is running? thanks, jrun

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?