Voicemail Server


#1

Am I correct in assuming this is possible?

Dial(SIP/${EXTEN}@voicemailserver.vm.domain.com)

How about this?

Dial(SIP/${EXTEN}@vm.domain.com)

voicemailserver is another asterisk box with the user provisioned in sip.conf, and relevant entries to route to voice mail on that box with extensions.conf and voicemail.conf. For the second example, I assume Asterisk can do SRV/NAPTR to look up the relevant host to use?

Is this routinely done? Is this the best way to separate out the voicemail from one asterisk system to another? Is SIP the best protocol for this, or would IAX be better? For SIP, how is the authentication done? Does the desatination server just ask for credentials and compare it against what it has in sip.conf (as if the first asterisk server was a client) ?

Oh, and this is important. When the Dial command is executed, does the first asterisk host stay involved in the RTP stream, or only the second?

Thanks.


#2

[quote=“dgarstang”]Am I correct in assuming this is possible?

Dial(SIP/${EXTEN}@voicemailserver.vm.domain.com)

How about this?

Dial(SIP/${EXTEN}@vm.domain.com)[/quote]

Yes, it is possible to network two Asterisk boxes in such a way.

Yes, it it makes sense to seperate services onto independent servers in order to provide scale. You may seperate into the PBX, voicemail server, IVR server, recording server, etc. Routinely done.

IAX2 would be better, as you may use IAX2 Trunking which is more efficient than SIP, details here:

voip-info.org/wiki-IAX+versus+SIP
voip-info.org/wiki/view/Aste … X+channels

You might also consider using DUNDi (dundi.com) to provide a federated PBX and make your services available in this fashion.

Depends on how your ‘canreinvite’ settings are set within your sip.conf and your notransfer in iax.conf.

voip-info.org/wiki-Asterisk+sip+canreinvite