Communication between two asterisk servers


I have two asterisk servers on two different networks, my first one delivers numbers from 100 to 105 and my second one from 200 to 205. How is it possible to call the 201 number from the 101 number phone?

If yes, what is the configuration file to be modified and with what variables?

Can you answer me in English and in French :wink:


J’ai deux serveurs asterisk sur deux réseau différents, mon premier délivre des numéros allant de 100 à 105 et mon deuxième allant de 200 à 205. Comment est il possible d’appeler le numéro 201 depuis le téléphone qui porte le numéro 101 ?

Si oui qu’elle est le fichier de configuration à modifier et avec quelles variables ?

Vous pouvez me répondre ne anglais et en français :wink:

You mean daemon, not server.

You almost certainly also meant to say using SIP, as this can also be done using analogue, PRI, ISDN, H.323, etc.

You define an endpoint for the other system, in pjsip.conf. Ideally do so by IP address, not registration (one has to be by address, in any case,) and, optionally add authentication. You handle outgoing calls (acting a SIP client), to the other instance, as you would an outbound call to an ITSP, and handle incoming calls (acting as a SIP server), as you would from a phone.

For worked examples, please hire a consultant.

Thanks for the quick answer,

No I have 2 servers: srv-voip-01 on the network and another on the network

I would like to know if it is possible to make a call from a 101 phone with an ip of 3.10 to a 90.10 phone with a 201 number

Merci pour la réponse rapide,

Non j’ai bien 2 serveurs : srv-voip-01 sur le réseau et un autre sur le réseau

et j’aimerais savoir si il est possible de passer un appel depuis un téléphone 101 qui a comme ip 3.10 vers un téléphone 90.10 avec comme numéro 201

Noting that server refers to the hardware, and not to Asterisk, and that there isn’t a need for the originating user agent to have an extension number (unique or otherwise), it is very easy if you have a basic understanding of how Asterisk configurations work.

A peer support forum like this is best used as a source of hints, or to help debug specific problems. It is not a substitute for courses, or paid services.

You could also consider using FreePBX, in which case the construct you want is an intra-company trunk.

merci je vais y jeter un coup d’œil

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