RE: [Asterisk-Users] audio message delivery

2005-05-25 Thread Tim Howell
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 > [EM

[Asterisk-Users] Small office setup with Asterisk @home, IAX and analog termination

2005-05-18 Thread Tim Howell
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 f

[Asterisk-Users] Asterisk @home with IAX termination...

2005-05-11 Thread Tim Howell
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 tri

[Asterisk-Users] Make voicemail use Maildir...

2005-04-01 Thread Tim Howell
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 del

RE: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme

2005-03-17 Thread Tim Howell
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 destination-

[Asterisk-Users] Cisco DTMF problem...

2005-03-15 Thread Tim Howell
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 DT

[Asterisk-Users] Incoming calls from Cisco 1760 given wrong context...

2005-03-15 Thread Tim Howell
ing for 711515 in from-sip-external 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;b