search for: planinternet

...pIdentifier: 12111 (0x2f4f) > 23/04/2004 09:27:02 } > 23/04/2004 09:27:02 SilenceSuppression: false > 23/04/2004 09:27:02 } > 23/04/2004 09:27:02 } > 23/04/2004 09:27:02 } > > tcpdump showed: > > 09:27:22.840015 modgud.dln.net.1026 > ipphone1.planinternet.net.12110: > udp 252 [tos 0x30] > > AquaGK recommended provider to use port 1028 for MediaChannel: > > 23/04/2004 09:27:02 Q.931, callee stream opened to 62.26.126.156:1720 > 23/04/2004 09:27:02 H.245, 0x08412500, send fast start elements to callee: > provider at 62.2...

Hi, how can I disable event_log in order to reduce hard disk activity? I can't find any hints in conf files. Must I hack the source code or even use brutal methods like creating a dir called event_log in the log dir, in order to prevent asterisk from creating an event_log file? (Just chmod a-w event_log does not work, unfortunately.) Thanks for any hints! Roger.

Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's "einen Ruf heranholen". It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to

I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323

Hi, some days ago we discused here the need for an analyzer for the 1000 Hz tone, as opposite application to Milliwatt. Here it is: Mwanalyze http://planinternet.net/download/voip/asterisk/app_mwanalyze.c It performs a Fourier analysis for a fixed frequency and tells the amplitude. The frequency is not limited to 1000 Hz, but can be passed as argument. The periode duration must be a mulitple of 0.5 ms, thus the valid frequences are: 2000 Hz, 1000 Hz, 666....

...t; Date: Fri, 27 Aug 2004 14:17:26 +0000 (UTC) > From: tony@softins.clara.co.uk (Tony Mountifield) > Subject: [Asterisk-Users] Re: how to fetch a call? > To: asterisk-users@lists.digium.com > Message-ID: <cgnfpm$56k$1@softins.clara.co.uk> > > In article <412F4122.6070401@planinternet.de>, > Roger Schreiter <roger@planinternet.de> wrote: > > Hi, > > > > there is a feature, which I would like to use with asterisk, > > and I assume it exists. > > Unfortunately I don't know how to say it in english. > > In german it's "e...

Hi, I have unpleasent short audio gaps when a perl based agi scripts starts. Thus, I now started to put all those things in C programmed daemons for fast-agi. Anyway I'm looking for another mean, which would help me more quickly. I noticed, that all agi scripts are running with system priority -11, like asterisk does. This is really waste of priority. I would like to have the AGI scripts

Hi, Digium's E400P has RJ45 conector and my E1 link has BNC concetor. Could someone tell me the cable pinouts to make this conection? thanks Eduardo

Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc.

Hi, I have to develop a phone application using asterisk's chan_oss. When the phone is idle, i.e. the last command was a hangup, one hears a "toot, toot, toot, ..." But unforuntaly its use is in Germany, where one expects a continous "toooooooooooooooooooooooooooooooooo ..." before dialing. Is there anything to define the tone indicating "ready to dial"?

Hi, I compiled asterisk and chan_h323 on an Opteron in 64 bit mode. In the h323's Makefile I replaced in line 24 CFLAGS += -march=$(shell uname -m) by CFLAGS += -march=k8 and also tried CFLAGS += -m64 -march=k8 Both solutions do compile, but when starting asterisk, a load error occurs: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi When I grep

Hi all, I have one asterisk server with one ISDN BRI connection to PSTN, with h.323 support (oh323) I buy some voip phones, and I connect them to the same switch as asterisk server is; all is at the same TCP network. I need to route some extensions from my DDI (DID) line at asterisk to some voip phones, and also to do: outgoing calls from any voip phone to PSTN via asterisk, and intermediate

Hi, > chan_oh323.so: undefined > symbol: __use_ast_pthread_create_instead__ is not a bug, it's a hint: use "ast_pthread_create" instead [what your were using] and means: replace in asterisk-oh/asterisk-driver/chan_oh323.c at line 3764 "pthread_create" by "ast_pthread_create" Roger.

Hi List, does somebody know a SIP Provider which offers reseller possibilities? Moritz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040828/bddb5898/attachment.htm

...t very easy to find in the internet, and maybe will almost disappear, because it is an "old" version, I put it on our little asterisk download page. Maybe it'll help someone. It works fine together with the other asterisk stuff (around version 1.0.7) located in that directory: http://planinternet.net/download/voip/asterisk Roger.

Hi, app_milliwatt is a nice tool for a quick check of the line quality. Anyway, hearing to that tone for more than a minute is painful. Does anyone know the "opposite" application, i.e. an application, that "hears" and listens for a 1000 Hz tone and displays the quality in any unit? If not, I'll think about developing one. Regards, Roger.

Hi, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes through the asterisk box in the middle. Using sip debug I even can't find

Hi, I'm looking for a mean to send digital data over an E1 line, just like isdn4linux or Capi via AVM's FritzCard is able to do it with BRI lines (e.g. for PPP or ISDN raw connections). I'm not looking for modulated audio data representing digital data, like fax or the analogue modems of former times. I want an interface to the ISDN raw data, with an outgoing call marked as

Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to

Hi, I'm using app_pppd with a Digium-PRI-card for PPP connections. I had some strange problems with some IP packets passing and some not, e.g. ftp login went well, but as soon as I tried to up- or download a file, noting was transferred. I finally guessed, it must have to do something with the packet size. Then I started pppd with the parameters mtu 296 and mru 296 as in further times with