Re: [Asterisk-Users] SNOM Hotdesking...

/bug_view_page.php?bug_id=6035 Maik Schmitt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Snom 360 record button?

(start recording during a call) and is configurable via Dial-Options. The only thing I don't know is how to activate it without sending the DTMF sequence. Maik Schmitt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] delay on pri dialling when asterisk is spliced between E1-Pri and legacy pbx

. where does this delay come from? has it to do with 'overlapdial=yes'? This is normal behaviour if you use '.' in your extensions.conf. Use '!' instead and Asterisk will start dialing immediately. Maik Schmitt ___ Asterisk-Users mailing list Asterisk-Users

Re: [Asterisk-Users] snom 220 hints lost after reload

://bugs.digium.com/bug_view_page.php?bug_id=0002522). Unfortunately it seems to be impossible to lower the re-SUBSCRIBE timeout of the snoms. :( -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP ___ Asterisk-Users mailing list Asterisk-Users

Re: [Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'

there is any way to turn it off. Just ignore it. The phones will still work. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] configuring md5 authentication

it should... Is there any extra setting that I have to define? Did you use the same realm, that was specified in the sip.conf? The default is asterisk. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP signature.asc Description: Digital signature

Re: [Asterisk-Users] snom soft phone

to test it but I will not install Windows just for testing a phone. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

. The hardware is a TE410P. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

Any ideas ? exten = _X.,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100 exten = _X.,50,Dial(Zap/g3d/${EXTEN}) exten = _X.,100,Dial(Zap/g3/${EXTEN}) -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

, RESTRICTED_DIGITAL, 31KAUDIO, 7KAUDIO or VIDEO. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

is there any special version from libpri or asterisk necessary since it works ? I'am runnig version: CVS-11/11/03-11:49:55 and it don't work :-( We have CVS-11/24/03-12:12:10 -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

Has somebody got it work at all ? I mean data calls (ISDN 64k) through asterisk. Yes. Works fine here with a PRI from DTAG and an Ascend. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: Asterisk European Tour: was RE: [Asterisk-Users] * Party in Paris

Not sure if this is real info or just a JPG that someone created. Is Stuttgart a definate date on the 30th? If so, where in Stuttgart?? These dates were just made up bye Rainer and me. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: [Asterisk-Users] SIP URIs and ENUM or other types of lookup

) Try !^\\+00393(.*)!sip:[EMAIL PROTECTED] and !^\\+(.*)!iax2:iaxtel/\\1!. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: [Asterisk-Users] A software FAX modem

was installed in /usr/lib. The newer Version was in /usr/local/lib. I recompiled the apps with LD_RUN_PATH set to /usr/local/lib and now it works. I've just send a fax over our pstn-sip router to our asterisk :) -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP

Re: [Asterisk-Users] A software FAX modem

, action=1) at pbx.c:1151 #11 0x0806380d in ast_pbx_run (c=0x8136550) at pbx.c:1635 #12 0x0806988e in pbx_thread (data=0x8136550) at pbx.c:1856 #13 0x400310ba in pthread_start_thread () from /lib/libpthread.so.0 Hope that helps -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp

Re: [Asterisk-Users] Check and restart script..

the server back up :-) Deactivate the cron job or make the prog check for a 'lock'-file... -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: [Asterisk-Users] Asterisk as a stand alone voice mail server (fwd)

try leaving yourself voicemail, first. That should give you a CiscoIPPhoneIconMenu instead of the CiscoIPPhoneText. _maybe_ the phone is happier with that... Still the same error. :( -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: [Asterisk-Users] Asterisk as a stand alone voice mail server

On Sat, Aug 09, 2003 at 11:04:49PM +0200, Siggi Langauf wrote: Hi again, this just popped into my eyes: On Fri, 8 Aug 2003, Maik Schmitt wrote: I just tried to use it with our 7960 (sip-version). I've set the services_url in SIPDefault.cnf to http://xxx.xxx.xxx.xxx/xmlservices/vm

Re: [Asterisk-Users] Asterisk as a stand alone voice mail server

://xxx.xxx.xxx.xxx/xmlservices/vm/index.php?user=1234amp;pin=1234amp;folder=INBOXamp;do=chfolder/URL Position4/Position /SoftKeyItem /CiscoIPPhoneText --- Any Ideas? -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature

Re: [Asterisk-Users] PBX / Asterisk integration

the existing PBX with Asterisk and SIP phones or channel banks. This will of course take some time. :) Another goal of the project is to route calls to other universities over IP if they support it. You can find some information about our project on our Website (see below). -- Maik Schmitthttp