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Hi all, Anyone knows of any Voice Recognition applications which use VoIP? Preferably open source. I am basically trying to build a Voice Call Router, something to recognize a spoken name and then transfer the caller to the right party's extension? TIA. -Eric -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages

We've been testing (aim:frziegler and aim:end1r) the Voicemail2 app for a few days now, based on a CVS build from Monday, 5/12/03-23:15. Works good! Thanks Mark! We seem to have found a bug in the MWI (Message Waiting Indicator) logic. By simply creating msg0000.txt files in both structures, e.g.: for extension 4000: voicemail1: /var/sp...

Hi all, Does anyone have an application/script or extensions.conf file which will do the following? "When a new VoiceMail is left for a user, the asterisk system will place a call to a cellphone/pstn number(via some provider). When the user answers his cell/home phone, comedian mail will ask for his password and he can check his Asterisk VM?" Anyone have any examples of it

Hi, I'm interested in what software (Free or course) that people use when they want to add a "dial by voice" service to their asterisk system. Meaning I pick up the phone.. dial some extension. it prompts me for name.. I say "John Smith".. and it dials his extension and connects the call.. TIA, -------------- next part -------------- An HTML attachment was

...at lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: A linksys SPA921 behind NAT and firewall (joakimsen at gmail.com) 2. Re: Making Asterisk a "Voice Router" (end1r) 3. Split asterisk in two ?? One TDM and One IP only?? (Steven) 4. Authenticate by IP? (Carlos Chavez) 5. Polycom 601 + Headset (Dovid B) 6. Re: tech prefix (Philipp Kempgen) 7. Re: Authenticate by IP? (joakimsen at gmail.com) 8. [France CID] Does Zaptel support ETSI FSK? (Vincent...

Hello all, I have a 7960 registered to asterisk. I am trying to use iconnect as my sip provider. When I send an invite to delta-three, I get the normal INVITE - 407 - INVITE exchange. The problem is, asterisk is sending the second invite using the 'dialed number' from the 7960 as the username, and not my 'username' configured in sip.conf. I believe that digest authentication

Hi all, I am trying to use canreinvite in sip.conf and park/pick up calls at the same time. Problem: When I have it set up so RTP goes through asterisk (sip.conf: canreinvite=yes), # to xfer works fine. But, when I set it up so the RTP goes direct between endpoints (sip.conf: canreinvite=no), the # to xfer doesn't work. I believe this is because asterisk isn't in the RTP path and