search for: steinwendtner

Hello ! Are there any plans at Digium to include also german voice prompts ? Thanks regards Hans

Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables

Hello ! I've upgraded our testsystem from asterisk 1.4.21 to asterisk 1.6.0.6. We 've noticed that the log files are now in colour. I could not find a note in the upgrade section about this. Is this a feature or a bug ? It might be usefull to have them not in colour. best regards Hans

Hello ! In order to chase after a problem I implemented the following dialplan to have an answertime of exactly one minute: exten => xxxxxxxxxxx,1,NoOp(Test wait) exten => xxxxxxxxxxx,n,Answer exten => xxxxxxxxxxx,n,NoOp(Current timestamp: ${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) exten => xxxxxxxxxxx,n,Wait(60) exten => xxxxxxxxxxx,n,NoOp(Current timestamp:

Hello ! I 'm using a TE405P with a HW echocanceller module attached on it. dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0. As far as I know, the fax tone detection is done on the FW board. How can I verify that the echo canceller has been turned off ? When I do a cat /proc/dahdi/1 for span 1 I see still the VPM450 entry near the channel although this was a fax call with CED tone.

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

Hello ! I 'd like to connect Cisco IP phones to *. (7940 & 7960) Shall I use SIP or SCCP. Which approach provides better support of features of the Cisco IP phones ? Thanks ! Johann

...r extensions?), reloads the extension.conf and so on. The 'reload' command do it all at once. Regards, ---- Filipe Mordhorst Joinville - SC - Brasil -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Em nome de Johann Steinwendtner Enviada em: sexta-feira, 24 de mar?o de 2006 08:49 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [Asterisk-Users] reload - restart Hi ! What is now the difference between a: reload - (cli command reload). restart - (I assume the application asterisk is restarted. o.k sta...

Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA

Hi all! I'm getting an error when I try to start asterisk with chan_misdn. I patched my kernel (2.6.12.4), and compiled the whole stuff (kernel, mISDNuser, asterisk, chan_misdn). I got mISDN from http://isdn.jolly.de/download/v3.0/ I'm using a CVS Snapshot of asterisk, which was checked out about 5 hours ago. This is the error: [chan_misdn.so]Aug 15 14:13:29 WARNING[4929]:

On Tue, Mar 21, 2017 at 09:36:21AM +0100, Olivier wrote: > Thanks to Tzafrir help, Now I have an Asterisk-Dahdi system installed from > Stretch repository. > This system has a Digium HX8 card with BRI modules, Asterisk 13.14.0, Dahdi > 2.11.1. > BRI spans appears as up and active but I've not tested them yet: > CLI> pri show spans > PRI span 1/0: Up, Active > PRI

Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator

Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1)

...0.6 To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <200903091110.43522.tilghman at mail.jeffandtilghman.com> Content-Type: text/plain; charset=3D"iso-8859-1" On Friday 06 March 2009 09:39:30 am Johann Steinwendtner wrote: > Sorry, that I wasn't clear enough. The logfiles contains escape codes = + > the colour codes. > e.g.: > [Feb 12 13:38:30] VERBOSE[19816] logger.c: =3D=3D Registered custom = function > 'ESC[1;36;40mSQL_ESCESC[0;37;40m' [Feb 12 13:38:30] VERBOSE[19816] > logg...

Hello ! Asterisk 1.0.9 running on Linux 2.6.12. I'm not able to call iax2 channels. There can be no translation path found. When I try to call from a ZAP PRI channel the following error occurs: channel.c:1891 ast_request: No translator path exists for channel type IAX2 (native 63488) to 72 dial_exec: Unable to create channel of type 'IAX2' What is wrong ? Here is my iax.conf:

Hello ! I 'm having a machine running asterisk 1.6.0.10 with IAX and dahdi. The calls are going in and out from IAX2 to dahdi (chan_dahdi + libpri) and vice versa. After a period of time, I got the following scenario: NOTICE[860] chan_iax2.c: I should never be called! WARNING[752] channel.c: Exception flag set on 'IAX2/iax-peer-13262', but no exception handler WARNING[752] channel.c: