Interconnecting with Asterisk

[Shane Harrison]

Hi there,

I would like to move to using SipXecs as my SIP based PBX at home. I
currently have an IP04 (Asterisk appliance blackfin box) with FX0 and FXS
ports.

What I am looking to do is to continue using the Asterisk box as a PSTN
gateway and to provide a couple of analog extensions. I would like however
for the voicemail for the analog extensions to be on the SipXecs system.

The PSTN gateway side of things is fine. My problem is more to do with the
analog extensions ie. how best to do this in a way that is of course
transparent to the end user. Initially I thought that the SipX should
simply be setup to send any calls for the analog extension to the Asterisk
box and when it times out, send the call to the voicemail. However I can't
have voicemail on the SipX without setting up a user and if I use the same
extension number eg. 300, then a call to 300 never reaches my dialplan
entry to send it to the Asterisk box since SipX sees a local user with
username 300 first.

Another way would be to have a dummy user on SipX eg. 399, and use that as
the Voicemail box for 300, but I suspect access to the voicemail box may
become more difficult for users.

A third way would be to get Asterisk to on forward the call after a timeout
on the analog extension and pass it directly to Freeswitch which has a
voicemail box for 300 that SipX doesn't know about. However again access
to the mailbox from the web portal has just got difficult again.

What I am really trying to do is set up Asterisk as an analog ATA I guess -
any pointers on the best topology to use would be appreciated.

Kind regards
Shane

[Todd Hodgen]

Shane, Your efforts to continue to use the Asterisk as an ATA, although
valiant, might be a daunting task. Have you considered simply picking up a
used Linksys ATA on ebay for $20-$30 and calling it a day? I suspect their
might be some resell value in your current IP04 to help pay for it. I
suspect this would make for a much simpler installation and much better use
of time. Especially considering most talk on this mailing list is around
commercial applications of this product.



From: sipx-users-***@list.sipfoundry.org
[mailto:sipx-users-***@list.sipfoundry.org] On Behalf Of Shane Harrison
Sent: Sunday, November 11, 2012 11:20 AM
To: sipx-***@list.sipfoundry.org
Subject: [sipx-users] Interconnecting with Asterisk



Hi there,

I would like to move to using SipXecs as my SIP based PBX at home. I
currently have an IP04 (Asterisk appliance blackfin box) with FX0 and FXS
ports.

What I am looking to do is to continue using the Asterisk box as a PSTN
gateway and to provide a couple of analog extensions. I would like however
for the voicemail for the analog extensions to be on the SipXecs system.

The PSTN gateway side of things is fine. My problem is more to do with the
analog extensions ie. how best to do this in a way that is of course
transparent to the end user. Initially I thought that the SipX should
simply be setup to send any calls for the analog extension to the Asterisk
box and when it times out, send the call to the voicemail. However I can't
have voicemail on the SipX without setting up a user and if I use the same
extension number eg. 300, then a call to 300 never reaches my dialplan entry
to send it to the Asterisk box since SipX sees a local user with username
300 first.

Another way would be to have a dummy user on SipX eg. 399, and use that as
the Voicemail box for 300, but I suspect access to the voicemail box may
become more difficult for users.

A third way would be to get Asterisk to on forward the call after a timeout
on the analog extension and pass it directly to Freeswitch which has a
voicemail box for 300 that SipX doesn't know about. However again access to
the mailbox from the web portal has just got difficult again.

What I am really trying to do is set up Asterisk as an analog ATA I guess -
any pointers on the best topology to use would be appreciated.

Kind regards
Shane