[HELP] How to install/configure asterisk as only SIP-proxy?


#1

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?


  1. I saw there’s 4 s/w at homepage, asterisk/zaptel/libpri/libiax, which

package I need to download/install ? I guessed only asterish, am I right?

  1. how can I configure asterisk? to make it run like a sip-proxy?

3x very much. I fully lost my voip path… :frowning:

F.


#2

[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?


  1. I saw there’s 4 s/w at homepage, asterisk/zaptel/libpri/libiax, which

package I need to download/install ? I guessed only asterish, am I right?

  1. how can I configure asterisk? to make it run like a sip-proxy?

3x very much. I fully lost my voip path… :frowning:

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.


#3

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


extensions.conf

[from-sip]
exten => 6270,1,Dial(SIP/6270,20)
exten => 6271,1,Dial(SIP/6271,20)



#4

Asterisk can operate that way too. It depends on the setting(s) of canreinvite= in sip.conf.


#5

[quote=“Telephony”]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


extensions.conf

[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.


#6

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


#7

[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.


#8

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


#9

[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.