similar to: SV: Running Asterisk and NAT on the same box?

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change 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

Hi I know this is a bit off topic, but still pretty interesting. I'm running Asterisk on my Linux router/NAT/FW connected via cable (1mbit/200kbit) to the internet. Now, I wanna do local QoS implementation. Just very simple to give RTP (UDP) highest priority on my outbound interface. So, whenever I got an ongoing call, the RTP traffic should be handled first and other data (file transfers

I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes.

Hi! Are the GIPS codecs 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 --------------------------------------

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:

My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r afoc@interconnessioni.it Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: SV: [Asterisk-Users] Queues

Hi! Does * do DNS-lookups when outgoing calls are placed to a different SIP domain? Can I call from <sip:1000@mydomain.com> to <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

OK! I'm sorry about this. As I wrote earlier I'm totally lost... but I will try to explain the problem in steps bellow, ok. 1. I installed the rpm's for samba, Version 2.0.2a-ssl I think this is the version distributed with redhat linux 7.0 2. Then I changed the parameters in the /etc/samba/smb.conf file, and in this file I added the folowing parameters. [global] netbios name

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

Sorry about that, its just outlook that does that by default. But manually deleted your adress now in reply. I don't know what you mean with "top posting"? What I mean is that if you have another security on the connection (be it physical security - the connection doesn't go over public means, or VPN - connection level encryption) then you don't need another encryption on

Ok, I'll post all the messages on the screen. -----Ursprungligt meddelande----- Fr?n: H. Peter Anvin [mailto:hpa at zytor.com] Skickat: fr 2003-05-30 22:21 Till: Olsson Lars Kopia: syslinux at zytor.com ?mne: Re: [syslinux] Problems with PXELINUX 2.05pre1 Olsson Lars wrote: > Reading config based on hw address is really what we need, so we quickly > downloaded

That was my impression as well, I tried adding switchtype=qsig in zapata.conf, but all I see is the capi information... No connection... Janne > -----Ursprungligt meddelande----- > Fr?n: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] F?r Peter Svensson > Skickat: den 23 februari 2005 21:34 > Till: Asterisk Users Mailing List -

-----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Terry Wade Skickat: den 19 augusti 2005 07:08 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' ?mne: RE: [Asterisk-Users] Zaphfc.ko module error Hi Remco Thanks for the response. I am running Suse 9.3 kernel 2.6.11.4-20a-default. Will check

Seems to me like the negotiation fails for some reason. Maybe you are trying to use a callerid that isn't allowed? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com genom Gary Richardson Skickat: on 2006-02-01 21:45 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: [Asterisk-Users] Re: CallerID Problem No, I'm not

How do you set the CallerID? Have you checked with your provider that they've enabled callerid? If yes, are you using a correct number that the provider allows? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Garth van Sittert Skickat: den 2 februari 2006 12:37 Till: Asterisk Users Mailing

http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com genom Brian J. Murrell Skickat: to 2006-02-02 20:14 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time I want Asterisk to delay answering the POTS line via a

Im curious. Does anyone have experienced echo-problems that later where solved by buying a hardware-echo canceller such as the Wildcard TE411P? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r James Harper Skickat: den 6 februari 2006 11:46 Till: Asterisk Users Mailing List - Non-Commercial

I don't quite follow you? There are USB headsets that don't require a soundcard at all. They have a built in soundcard which (I suppose) could be better than the crap they build into most laptops. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r El Flynn Skickat: den 24 maj 2006 10:17 Till:

Hi Chris, Yes the call was from PSTN and your solution worked great! I've read about SetCallerPres earlier but I didn't connect the dots this time. Thanks alot! :) Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Christoph F?rstaller Skickat: den 29 mars 2007 15:29 Till: Asterisk Users

This is not a job for dovecot. You should look into whatever is your MTA (exim, postfix etc) and implement the solution there. But my initial suggestion is to check SPF and DKIM of the email. Because I know that gmail does terminate spammers quick, but if you don't validate SPF or DKIM, you might be a victim of spoofed Gmail email. Best regards, Sebastian Nielsen -----Ursprungligt