hi,
I am a freshman on voip/asterisk. I want to setup a SIP-proxy-server
on my linux box. (don’t have any PSTN card in my box), that is to say,
I want to let my linux accept ip-phone-request and pass to next SIP
server, how can I do?
package I need to download/install ? I guessed only asterish, am I right?
3x very much. I fully lost my voip path… 
F.
[quote=“fulcrum”]hi,
I am a freshman on voip/asterisk. I want to setup a SIP-proxy-server
on my linux box. (don’t have any PSTN card in my box), that is to say,
I want to let my linux accept ip-phone-request and pass to next SIP
server, how can I do?
package I need to download/install ? I guessed only asterish, am I right?
3x very much. I fully lost my voip path…
F.[/quote]
When it comes to SIP, there are 2 relevant Asterisk files: sip.conf and extensions.conf. Simplistically speaking, sip.conf takes care of the registration issue and extensions.conf is used for the actual calls.
Unlike [Open]SER which handles the text aspect of the call and allows 2 SIP devices to talk (via codecs) directly to each other, Asterisk has to be in the middle of the call.
I believe the following to be true, but I’d love to hear from someone that really kows what he’s talking about.
if you are in the middle of a SIP session started with SIP Express Router and the server dies, your call will continue uninterrupted.
On the other hand, when the call is sponsored by Asterisk, if the Asterisk server dies your call will come to a halt.
-RFH
ps: see my SIP settings in my next posting.
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip
[6270]
type = friend
username = laurel
password = secret
callerid = “El Phon de Ramon” <6270>
host = dynamic
nat = yes
[6271]
type = friend
username = hardy
password = anothersecret
callerid = “El Fon de Ramon” <6271>
host = dynamic
nat = yes
[from-sip]
exten => 6270,1,Dial(SIP/6270,20)
exten => 6271,1,Dial(SIP/6271,20)
Asterisk can operate that way too. It depends on the setting(s) of canreinvite= in sip.conf.
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip
[6270]
type = friend
username = laurel
password = secret
callerid = “El Phon de Ramon” <6270>
host = dynamic
nat = yes
[6271]
type = friend
username = hardy
password = anothersecret
callerid = “El Fon de Ramon” <6271>
host = dynamic
nat = yes
[from-sip]
exten => 6270,1,Dial(SIP/6270,20)
exten => 6271,1,Dial(SIP/6271,20)
-----------------------------------------[/quote]
hi, telephony, 3x for ur help.
you know, I need to setup an sip-proxy server, which need to conect a
sip-register server, but in your sip.conf, I didn’t find where to setup the
ip/port of the sig-register server, could u help?
(by the way, u also agree that I need only asterish, no need zapaca?)
F.
Hi fulcrum:
If you need to separate the SIP server funcionality (registration, proxy, etc.) you shouldn’t use Asterisk. Since * offers such a unified treatment (SIP, POTS, etc., etc.) of telephony technologies, it also hides and “bundles” some of the inside details. With Asterisk you are getting a heck of a lot more than SIP service.
To get a SIP server (and only a SIP server) you should look into the SIP Express Router, which comes in two almost identical versions: ‘ser’ and ‘openser’ because their developers had a fight and parted ways.
You must be asking about “Zapata” (not Zapaca). If that’s the case, you can get started without telephony hardware BUT you MUST get a voice card of some sort (Digium is the path of least resistance) when you go into actual use and production.
-RFH
[quote=“Telephony”]Hi fulcrum:
If you need to separate the SIP server funcionality (registration, proxy, etc.) you shouldn’t use Asterisk. Since * offers such a unified treatment (SIP, POTS, etc., etc.) of telephony technologies, it also hides and “bundles” some of the inside details. With Asterisk you are getting a heck of a lot more than SIP service.
To get a SIP server (and only a SIP server) you should look into the SIP Express Router, which comes in two almost identical versions: ‘ser’ and ‘openser’ because their developers had a fight and parted ways.
You must be asking about “Zapata” (not Zapaca). If that’s the case, you can get started without telephony hardware BUT you MUST get a voice card of some sort (Digium is the path of least resistance) when you go into actual use and production.
-RFH[/quote]
hehe, RFH,
sorry for the typo - zapaca, haha.
you know, I knew * is a unified/integrated s/w, and currently I need only
its SIP capablity, but in the future I will use more of its function.
so, how can I set my * to connect other SIP server ? (then, when I dial
into my * box, when press “2”, the * will connect into other SIP server
with the pre-define user/pass.)
3x.
F.
I am going to defer this to the more qualified folks out there.
So far I have SIP inter-extensions figured out, and that’s about it.
Hey, I am a freshman, too (but growing by leaps and bounds)!
Have you tried the Google Asterisk-users group? It has a lot more traffic than this one. Another resource is the mailing list:
http://lists.digium.com/mailman/listinfo/asterisk-users
-RFH
[quote=“fulcrum”][quote=“Telephony”]so, how can I set my * to connect other SIP server ? (then, when I dial
into my * box, when press “2”, the * will connect into other SIP server
with the pre-define user/pass.)[/quote][/quote]
You can’t. What would you expect to happen if you could?
You can call a number on the other SIP server with a normal SIP dial command in extensions.conf.
Asterisk is a telephone exchange. What you’re asking is the equivalent of asking how you can call your local (PSTN) telephone exchange but not call a specific number at that exchange - it doesn’t make any sense. What phone do you expect to ring? The operator? If so, you have to call the specific number for the operator.