search for: dtfmmode

Displaying 4 results from an estimated 4 matches for "dtfmmode".

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2005 Feb 17
1
(Kphone) Registration Failed: Forbidden
...exten => 8003,3,Voicemail(u8003) exten => 8003,103,Voicemail(b8003) exten => 8003,104,Hangup And in sip.conf i have [8003] type=friend host=dynamic ;<- This is supposed to allow registration, isnt it? callerid=Ariel Molina <8003> mailbox=8003 dtfmmode=info username=ariel secret=wow disallow=all allow=ilbc allow=gsm allow=ulaw allow=alaw allow=g723.1 allow=g729 However when i try to register with kphone i get a dialog asking for my passwd. I give it and then i get (Kphone) Registration Failed: Forbidden I k...
2005 Mar 20
1
Problem transfering incoming calls
Guys. Im having a big problem transfering incoming calls thru zap channels to some other extension. If the call is made by me to the outside via zap channels, no problem, hitting # gets me the transfer prompt, but if the call comes in thru zap and eventhough I am sending the call from the zap channel to my sip ata (GS ata 286) using Dial with wtWT as parameters, when hitting # I don't hear
2005 May 31
0
Receive calls with Aastra 480i phone problem
...this: the phone is able to place calls. But as soon as the phone receives a call, you are able to answer it. But then, 2-3 seconds later, the call is dropped. The extension works, because I've put his Gnet phone back until I figure out what is wrong with this phone. I tried changing the dtfmmode for that particular SIP extention. To no avail. I'm running out of idea, now. Anyone ever configured those phones? Cheers, -- Jean-Francois Theroux Systems administrator PrivalODC 450.761.9973 ext 503 http://www.privalodc.com
2017 May 22
3
SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)
Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the