search for: staffan

...ot;, but when plotted, the figure doesn't look like the partial prediction for "dayno". Instead, it gives me a very scattered figure ("shotgun effect"). Has anyone tried to extract the partial predictions? If so, please could you advise me how to do this? Thanks, Staffan P.S.. For comparison, please have a look at Simon Woods paper in R News, 1(2):20-25, June 2001, especially the figures showing the partial predictions of Mackerel egg densities. Using those graphs as an example, I would like to extract the partial predictions for e.g. "s(b.depth)",...

...e the default 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...

...<sip: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

...odecs now implemented with the Asterisk? If I need more analog lines, say around 30, what'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

...risk crashes. 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

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

...ult thing to explain with a language that is not mine :D I shut off the PDC on all the vlans, and it is only running on lan0 : joining domain works perfectly. Thank everyone who has read this ;) Val?ry P.S. : the previous thread I initaited is not finished. I must give the solution, given by Staffan Emr?n (thank you very much Staffan). The fact is the rights are given before login time. So even if I write a script to put a domain user/group in the local administrators group, it won't have the rights until next connection. For adding the user to the local administrators, I'm using C...

...unning on 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 et...

...and other 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 d...

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: asterisk-users@lists.digium.com ?mne: RE: [Asterisk-Users] Mailing list growth What about splitting asterisk-users in 2 groups asterisk-tdm and asterisk-voip -----Men...

...HRD - ERRgeneral (General failure.) SMB connection failed If anyone knows about this error, and how to resolve it, I will be most grateful. I have googled on the error message, and found a german page with a solution (add the workgroup= option), but that didn't work out for me. Best regards, Staffan Emren Societas Archaeologica Upsaliensis http://www.sau.se