search for: kerker

...fault SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this up and running? SER/Asterisk on the same machine? rgds, /Staffan kerker -----Ursprungligt meddelande----- Fr?n: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Skickat: den 6 november 2003 12:45 Till: asterisk-users@lists.digium.com ?mne: Re: [Asterisk-Users] How to control dialout in extensions file You could use DISA app. exten => 2101,1,DISA,/opt/pass.txt|defaul...

...2000@remote_domain.com>? Can * work as a regular SIP proxy in that aspect? Can * handle SIP URI:s that are complete SIP URI:s (sip:user@domain) instead of numbers only? Or should I run a SIP proxy on a different machine to handle pure SIP requests and let * register an account? rgds, /staffan kerker sweden _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

...shes. When I debug sip I get a "noisy feedback" from SER, and then asterisk crashes. the only debug information asterisk is leaving is "segmentation fault, dumping core". anyone got a clue? i'm not running the latest CVS so maybe i should upgrade first... rgds, /staffan kerker -----Ursprungligt meddelande----- Fr?n: firedude@shorelinuxsolutions.com [mailto:firedude@shorelinuxsolutions.com] Skickat: den 30 november 2003 20:04 Till: asterisk-users@lists.digium.com ?mne: Re: [Asterisk-Users] asterisk server crashing

...;s the easiest way doing it? I checked the Mediatrix box with 24 connections, maybe that would be a good (and rel. cheap) way to go? Any other suggestions? The ports has to support fax machines. rgds, /staffan -------------------------------------- -------------------------------------- Staffan Kerker / KIT Communications, AerotechTelub mail: staffan.kerker@aerotechtelub.se "Don't get involved in politics man, just play the gig..." /Sgt. Floyd, Electric Mayhem Band

Hi Is there any work being done on implementing IM/SIMPLE support for SIP on Asterisk? Like a presence server? rdgs, /Staffan Kerker

...n my FW/NAT/Router with two interfaces. My local phones are situated behind the NAT and connects to the outer interface of the */FW/NAT/Router. * is then connected to my SIP providers (since I'm only using the SIP-part of *, PSTN connection through my SIP-provider). Works fine! rgds, /staffan kerker sweden -----Ursprungligt meddelande----- Fr?n: Steven M. Sokol [mailto:ssokol@sokol-associates.com] Skickat: den 22 oktober 2003 17:22 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] Running Asterisk and NAT on the same box? Has anyone tried installing * on a box with two eth inter...

...r data (file transfers that may be ongoing) should back off. I've done this very simple with Cisco IOS routers by just using the priority queueing... but how do I do this (the easiest way) on a RH9 machine? Don't need any classes more than 1. priority RTP 2. all other data. rgds, /staffan kerker -----Ursprungligt meddelande----- Fr?n: B?asse Christophe [mailto:beasse.ch@wanadoo.fr] Skickat: den 31 oktober 2003 09:54 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] Password in VoiceMail Hi, In voicemail. i declare : 1000 => 1234, xxxx yyyy,tintin@milou.com I don'...

Hi Isn't this exactly what we _don't_ wanna do?! =) I suppose TDM and VoIP is supposed to interconnect not to be separated. i think it's nice with a busy list, it means some real hot stuff is happening, and that's good! rgds /staffan -----Ursprungligt meddelande----- Fr?n: Luciano Ramos [mailto:lramos@telviso.com.ar] Skickat: den 9 januari 2004 14:12 Till: