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Anna Crepes: Traubenzucker + Feldsalat spezielles Dressing (bringt selbst mit?) -------- Weitergeleitete Nachricht -------- Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Thu, 11 Dec 2014 15:34:39 +0100 Von: Markus <universe at truemetal.org> An: universe at truemetal.org Geschenke Moritz: dunkle Schokolade. Geschenke Anna: normale Schokolade. -------- Weitergeleitete Nachricht -------- Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Wed, 10 Dec 2014 05:12:16 +0100 Von: Markus <universe at truemetal.org> An: univer...

...schrieb Markus: > Anna Crepes: Traubenzucker > + Feldsalat spezielles Dressing (bringt selbst mit?) > > > > -------- Weitergeleitete Nachricht -------- > Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. > Datum: Thu, 11 Dec 2014 15:34:39 +0100 > Von: Markus <universe at truemetal.org> > An: universe at truemetal.org > > Geschenke Moritz: dunkle Schokolade. > Geschenke Anna: normale Schokolade. > > > > -------- Weitergeleitete Nachricht -------- > Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. > Datum: Wed, 10 Dec 2014 05:12:16 +0100 > Von:...

.... I think the cool part is that it really tries to get the most accurate coordinates for every destination, down to the city level, by using text (country code, ISO, lat/long) and MySQL (ITU self-compiled DB + MaxMind public DB) based databases, which are included. Here are some demos: http://truemetal.org/universe/voipgmap/live_demo1.html http://truemetal.org/universe/voipgmap/live_demo2.html http://truemetal.org/universe/voipgmap/live_demo3.html http://truemetal.org/universe/voipgmap/live_demo4.html First I thought about calling it AstGMap but then realized you can simply feed it any form o...

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

Markus <universe at truemetal.org> schrieb: > I don't think so. Most users will use the router provided by Telekom. These users do NOT use Asterisk on theis Telekom-line... I asked for someone using Asterisk on Magenta Zuhause... :) > Anyway, after 15 seconds of Google'ing for Magenta Zuhause and SIP, &gt...

hi, today i upgraded from asterisk 1.4.21 to 1.6.2.9 (i know this release is not supported anymore, please don't tell me to upgrade). unfortunately now i can't use the rxfax() application anymore. i tried to use ReceiveFAX() the way i used to use rxfax() (multiple dedicated fax extensions getting their faxes from isdn via zaptel/dahdi), but this doesn't work. either fax

Hi list! I'm trying to make a SIP test call from Bria and/or 3CXPhone from a PC behind NAT. From Bria/3CXPhone I connect to an Asterisk 11.25.0 server on the internet at 100.100.94.210 with a SIP account "3333" created in sip.conf: [3333] type=friend secret=something host=dynamic nat=yes qualify=no disallow=all allow=alaw allow=ulaw canreinvite=no context=voipin I dial +1234

Hi list, I have a fresh install of Asterisk 12.0.0 and I'm going to use it only as a client. I'm trying to SIP REGISTER with a remote SIP provider. The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of the VM performs static NAT from the RFC IP address to a dedicated public IP address, however, they are rewriting ports at will.

Hi list! I'm getting this error frequently: ERROR[25193][C-0004f387]: cdr_mysql.c:203 mysql_log: Cannot connect to database server localhost: (2026) SSL connection error: SSL_CTX_set_default_verify_paths failed Right now, as a workaround, I reload Asterisk via cron once an hour, and after the reload everything is fine again _for a while_. Still, over the course of a month I lose about

Hi list! I think there are many german users in this ML, that use Asterisk with the new line of Deutsche Telekom (Magenta Zuhause). My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right now I can just hope, that I configured my Asterisk well to work with Deutsche Telekom, but I cannot be sure, since I can't test it... So my question: can someone using Asterisk with

I am trying to block/hide outgoing caller id on a PRI E1. It seems that it should be fairly simple, but it is defeating me. https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says: "to hide your caller id, use: Set(CALLERID(num-pres)=prohib)" That doesn't seem to do it. The calls are originated from AMI and I have tried a blank "CallerId:" line and

hi, We have a phone number from third party provider which is used for inbound calls. How could I monitor if this *phone number* is reachable? the initial idea doesn't sound elegant: - on my SIP server I set couple seconds of ringing before Answer(). - the monitoring server calls to that phone number for few seconds, checks if it "hears" the ringing and hangs up the call. ** I use

Am 13.06.2015 um 13:54 schrieb Luca Bertoncello: > I think there are many german users in this ML, that use Asterisk with the > new line of Deutsche Telekom (Magenta Zuhause). I don't think so. Most users will use the router provided by Telekom. Anyway, after 15 seconds of Google'ing for Magenta Zuhause and SIP, maybe this will help you:

Help! I have providers configured that send me incoming calls via SIP. There's one that seems to make trouble. As soon as I get a few concurrent calls through this peer, Asterisk CPU load goes up to 250%, audio becomes laggy and I get hundreds of these per second in the logs: [Jun 20 20:00:16] WARNING[1206][C-00000001] channel.c: Exceptionally long queue length queuing to

Hi list! Is the following application possible with Asterisk with the default functionality that is currently implemented? 1. User dials in via PSTN / SIP. 2. System announces random but individual PIN to user. 3. If there are no other users, user gets put through to a party by Asterisk dialing out via SIP and the user gets connected to that person. Let's call that person callee. 4. If

Hi list, I've always about 50 concurrent SIP callers listening to several MOH streams (fed via mpg123) on Asterisk 11.4.0, 4x 2.2 GHz and the CPU usage is always at about 250% causing stuttering of the streams and delays when using Read() and Playback() (overall everything is just really slow/delayed). I've seen Asterisk doing that when the MOH stream that was fed via mpg123 was

Dear list, maybe not really an Asterisk question, but... all my users dial in via PSTN (via SIP DIDs) and enter a target number via DTMF through my Asterisk 1.4. Out of about 150,000 calls per month I see on average about 1 call per month where an arbitrary caller enters the letter 'A' via DTMF. These callers use their mobile phones to dial in. I just reread the Wikipedia article on

Hi list, probably this is a FAQ but I can't seem to find it. How to find the RTP IP address of an ongoing SIP call? "sip show channels" seems to list the RTP address under the very left column called "Peer". And it also lists the associated "Call ID" which I could associate with a call by executing sip show channel <Call ID> and before figuring out the

Hi, has the following been done before respectively is it possible with Asterisk? I searched the archives but couldn't locate anything. 1. Call to 5555 comes in via SIP. 2. Call is not answered yet but progress continues. 3. At the moment the call comes in something like this gets spawned in the background: Dial(SIP/123456 at provider,,D(ww${EXTEN}) which should translate to:

Hi list! When I receive an incoming call from a SIP peer where I've configured disallow=all allow=alaw (and no other codec) I can see the following NOTICE on the console: Dropping incompatible voice frame SIP/peer07-0000007c of format ulaw since our native format has changed to (alaw) My question is: where can I change the native format from ulaw to alaw (or something else)? Is ulaw, as