asterisk users - Oct 2011 - Keeping Voice Call Active During Data Connectivity Loss

Greetings-

I'm working on a unique Asterisk installation where I've been given a
requirement of keeping a voice call active, even during a data connectivity
loss. So, let's assume I have remote users connecting to an Asterisk server
via "sometimes unreliable" connectivity such as satellite, wireless,
or <shudder> dial-up. It is certainly possibly this connectivity will go
down for a period of time anywhere from a few seconds to a few minutes (or
more). During this outage, if a call was already in session, is there any way to
prevent the call from be hung up, and simply kept alive until media can begin
flowing again?

In this situation, both sides of the link would be running Asterisk, 1.4.x or
1.8.x. Is this as simple as telling both sides not to hangup at a lack of media?
Are the steps the same whether using SIP or IAX (preferred IAX in this usage
case, unless SIP is specifically required)?

Thanks!

--Tim

Assuming that you don't have some sort of "reconnect protocol"
going on like
SIP headers,  a native-bridge to a local channel might do the trick for you.
If you are using DAHDI, you might be out of luck.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, October 03, 2011 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Keeping Voice Call Active During Data Connectivity
Loss

Greetings-

I'm working on a unique Asterisk installation where I've been given a
requirement of keeping a voice call active, even during a data connectivity
loss. So, let's assume I have remote users connecting to an Asterisk server
via "sometimes unreliable" connectivity such as satellite, wireless,
or
<shudder> dial-up. It is certainly possibly this connectivity will go down
for a period of time anywhere from a few seconds to a few minutes (or more).
During this outage, if a call was already in session, is there any way to
prevent the call from be hung up, and simply kept alive until media can
begin flowing again?

In this situation, both sides of the link would be running Asterisk, 1.4.x
or 1.8.x. Is this as simple as telling both sides not to hangup at a lack of
media? Are the steps the same whether using SIP or IAX (preferred IAX in
this usage case, unless SIP is specifically required)?

Thanks!

--Tim

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Maybe you could use a very simple sollution like a meetme room - you have
only to be "creative" with the dialplan.
Ioan
www.modulo.ro
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