hi all, how to establish a call between two asterisk servers for the sip users registered for the servers.
----- Original Message ----- From: <[EMAIL PROTECTED]> To: <[email protected]> Sent: Sunday, February 10, 2008 11:30 PM Subject: asterisk-users Digest, Vol 43, Issue 30 > Send asterisk-users mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. Re: Domainname for outgoing uri-dialing (B. Haje) > 2. Re: oneway audio with asterisk behind cisco pix 506 (Adam KOSA) > 3. Re: Asterisk Scalability (Bryan M. Johns) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 10 Feb 2008 18:11:01 +0100 > From: "B. Haje" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Domainname for outgoing uri-dialing > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > [EMAIL PROTECTED] wrote: >> 8 feb 2008 kl. 13.24 skrev Bjoern Haje: >> >>> Hi, >>> >>> I use outgoing URI-dialing for my sip-phones as suggested in >>> http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial >>> >>> The relevant extensions look like this: >>> >>> [dial-uri] >>> exten => _[a-z].,1,Macro(uridial,[EMAIL PROTECTED]) >>> exten => _[A-Z].,1,Macro(uridial,[EMAIL PROTECTED]) >>> exten => _X.,1,Macro(uridial,[EMAIL PROTECTED]) >>> >>> [macro-uridial] >>> exten => s,1,Set(dialuri=${CUT(ARG1,\;,1)}) >>> exten => s,n,Set(CALLERID(number)=${CALLERID(number)[EMAIL PROTECTED]) >>> >>> exten => s,n,Dial(SIP/${dialuri},120,tr) >>> exten => s,n,Congestion() >>> >>> I end up with an outgoing SIP-Invite with contact and from-headers >>> like [EMAIL PROTECTED]@<IP-address> >>> >>> That obviously is not what I want. I can set the fromdomain value in >>> the general-part of my sip.conf and leave away the setting of the >>> callerid which fixes the problem. But as I want to use different >>> domains for the outgoing calls depending on the user, that is not a >>> solution for me. Can I influence the generation of the outgoing >>> domainname somehow? >> >> No, but that would be a good addition to Asterisk. I started >> experimenting with that in my caller ID utf8 branch at some point, >> but never got time or funding to complete that work. > > Thanks for your help again. Would be nice really, but I'll try to find a > workaround to avoid that problem (or ignore it). > > Bjoern > > > > > ------------------------------ > > Message: 2 > Date: Sun, 10 Feb 2008 18:44:46 +0100 > From: Adam KOSA <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco > pix 506 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >> permit udp any host 192.168.5.0 range 10000 20000 and then I didn't > > home users typically use /24 netmask. If this is the case, i don't > understand why do you write keyword host following a network address. > > either specify a valid host address, or write 192.168.5.0 255.255.255.0 > to specify the whole subnet. > > if the netmask isn't /24 then, of course the above 5.0 may be a valid > host address. > > regards > adam > > > > ------------------------------ > > Message: 3 > Date: Sun, 10 Feb 2008 12:54:44 -0500 > From: "Bryan M. Johns" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Asterisk Scalability > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > We have multiple installs that tested-out at nearly concurrent 400 SIP > channels on a Dell 2950 with 2Xquad core at 1.6 Ghz, 16 GB of RAM. > > Bryan M. Johns > Shelton | Johns > Office: 678.248.2637 > FindMe: 678.229.1809 > Support: [EMAIL PROTECTED] > http://www.sheltonjohns.com > > On Feb 8, 2008, at 5:09 AM, Femi wrote: > >> Hi, >> Does anyone have data on the switching capacity of Asterisk based on >> the >> hardware? >> I need to know what type of hardware would be required to switch 100 >> simultaneous calls as opposed to 1000 or 10000 calls, no TDM just >> SIP to SIP >> VoIP calls >> >> Thanks >> >> Femi >> >> >> >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 43, Issue 30 > ********************************************** > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
