Darren Wiebe wrote:
I've been looking at a similar problem. Mine is slightly different
but it involves a customer phoning in, leaving a recording, and
having that recording delivered to a list of users. I hope to code
at least some of this in the next few weeks.
Darren Wiebe
[EMAIL
I'm setting up a small office with about 8 SIP phones. Incoming and
outgoing lines will be through IAX. We would also like to use an analog
line for 911. Is the TDM01B a good option for this kind of
configuration? Are there gotchas I'm missing?
Finally, we would like to be able to use analog
I've got an Asterisk @home setup that mostly works. I'm using Polycom
SIP phones and have PSTN termination through an IAX provider. I say
mostly works because, if I'm on the phone, any additional incoming
call goes straight to voicemail rather than signaling call waiting on
the phone. I've
What would be involved in modifying the Asterisk voicemail system to
use Maildir for storage? My thinking is that if voicemail were
delivered to Maildir it would be easy to configure an IMAP server to
work with Asterisk, and that would provide pretty good voicemail/email
integration (where
Gilbert Abboud wrote:
I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk
through SIP. Can you please send me the Dial-peer configuration that
creates a trunk between the Cisco router and Asterisk.
You can try something like this:
dial-peer voice 900 voip
Shouldn't it match Cisco1760_mc?
I've included the full debug below.
Thanks in advance for your help. I'm happy to provide any additional
information.
--TWH
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK1E6D
From: Tim Howell sip:[EMAIL
I've recently setup Asterisk. Calls are routed to Asterisk through a
Cisco 1760 router. Some calls originate at Cisco 7960 phones connected
to the router, some originate at other phones that are switched by a
legacy PBX.
My problem is that calls that begin at the 7960s do not seem to transmit
