, or if
there really are some conflicts with SuSE that may cause audio
instability. If so then it definitely would explain a lot regarding my
poor audio quality problems.
I would be happy to hear thoughts that any of you might have.
Thanks so much!
Darren Nay
[EMAIL PROTECTED
Just out of curiosity. Would you mind sharing that app_rxfax.c file
that you modified to work with SpanDSP 0.0.3?
TIA,
Darren
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ardjan
Zwartjes
Sent: Monday, January 22, 2007 2:12 AM
To:
) and
then discontinue the audio file stream and bridge the calls once an
outbound call is connected.
Can anyone point me in the right direction on how to do this with * ?
Thanks in advance,
Darren Nay
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Sounds like you need to dig into the documentation for the 7970 and
perhaps even contact Cisco TAC if that doesn't help.
It doesn't sound like your problem is related to Asterisk. The Cisco IP
phone won't register with asterisk until it's been provisioned. Those
7900 series cisco phones
of the rxfax
application and never writes the fax.tiff file.
Has anyone seen this behavior before? Any help that you could provide
would be very much appreciated.
Thanks in advance!
Darren Nay
[EMAIL PROTECTED]
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Nay
From: Darren Nay
Sent: Tuesday, January 16, 2007 10:25 AM
To: 'asterisk-users@lists.digium.com'
Subject: Asterisk, SpanDSP and RXFax
Hey All,
I am attempting to get the RXFax app working and having a hell of a time
of it. I am hoping that some
I just need the calling number
available somewhere ... I can easily use an AGI script to parse it out of a
string and pass it to VoiceMailMain .. I just need access to it from an AGI
script in order to do that.
Any ideas?
Thanks for any help you can provide!
Darren Nay
Ionosphere, Inc.
Vo
Sipura does include STUN as an option. It has for quite some time. We are
using it with all of our Sipuras behind NAT'd gateways and it works great!
Try upgrading to the latest Sipura firmware rev.
Darren Nay
-Original Message-
From: John Todd [mailto:[EMAIL PROTECTED]
Sent
ive method for this.
However, if we can use a mysql database for voicemail storage then I can set up
mysql database replication and our problem is solved.
Thanks so much for your help!
Darren Nay - [EMAIL PROTECTED]
that it will automatically detect the
number and then go directly to asking them to input their password.
If so, where would I make the config changes for this in the
asterisk config files, and does anyone have an example of a similar config?
Thanks!
Darren Nay
VOIP Network Developer
via a MySQL database?
Thanks for the help!!
Regards,
Darren Nay
[EMAIL PROTECTED]
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