[Asterisk-Users] Broadvoice's changes last week broke callforwarding

[Marios Andreou]

Oopps, 
sorry Paul I didn't understand your  issue, that's for sure. :(

Hmm Interesting thing though.
I'll try it and I'll let you know.

Although how can you reinvite a PSTN line?
They probably have canreinvite=no or similar (because they are not using *) for billing purposes.
If there is a reinvite from a PSTN/SIP line to your * then a transfer to another PSTN/SIP line and connect the 2 you are out of the
picture and in the CDR it will be ????? The CDR guys will probably know the answer to this. 

When you look at your Account online you see that you placed a call out?
If you get out of the picture (canreinvite=yes) then what will be the time of your phone call?

This is interesting.

I'll try it with a PSTN -> Broadvoice# -> * -> PSTN and PSTN ->Broadvoice# -> * -> SIP.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul Nuyujukian
Sent: Tuesday, March 15, 2005 2:48 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Broadvoice's changes last week broke callforwarding

Marios,

I don't think you quite understand my issue. The ata in my apartment is
behind a nat, and it always has had canreinvite=no. But my question
deals not with my sipura device, but calls entirely contained within the
* server (which again, is a live IP machine). A call comes in from
broadvoice to the * server, then the * server tries to forward that call
out through broadvoice and out to another number, a la
Dial(SIP/broadvoice/##########). There should be no need for
canreinvite=no set for the broadvoice peer, but the call forwarding will
not work unless reinvite is disabled. There is no issue with nat or
anything here, entirely internal routing, but for some reason, it does
not work.

If anyone has some time, can they try this and see if they have a
similar error? The only condition is that the * server will need to be
on a live IP.

set canreinvite=yes for the broadvoice peer.
setup your dial plan such that an incoming broadvoice call is answered
and then forwarded to a PTSN number through broadvoice again.
In my case, this does not work unless I set canreinvite=no for the
broadvoice peer. Does anyone experience the same issue?
I'd like to know if there is a problem on my side or if this is just a
complication of the new changes broadvoice made.
Perhaps someone knows why this is happening anyhow, without having to
test it. All thoughts are welcome.

I'm not sure if this post will reply properly to the thread (my last one
didn't) so just in case, links to the previous threads are below:
http://lists.digium.com/pipermail/asterisk-users/2005-March/094744.html
http://lists.digium.com/pipermail/asterisk-users/2005-March/094812.html


Paul

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