search for: crisco

> On a related note, how do you get a Cisco 7940 to dial numbers with a > hash in them, instead of just using the hash as a dial key. For > example, I have *#21# to check diverts, but the phone will just dial > "*" as soon as you type the # after it. <DIALTEMPLATE> <TEMPLATE MATCH="#..." Timeout="5" User="Phone" /> <TEMPLATE

...m an internal sipura spa-x000 (at extn 141), it rings straight through to extn 666. using the same dialplan, from a cisco 7960 with 7.1 sip code (at extn 142), i have to go off hook hit NewCall punch 142 (or any valid extn in the dialplan) hit Dial then dial 666 wtf? sip.conf for crisco [fiji] callerid="crisco" <142> type=friend host=dynamic port=5060 secret=pfui qualify=1000 dtmfmode=rfc2833 canreinvite=yes context=in-internal extensions.conf [in-internal] exten => s,1,Answer exten => 141,1,GoTo(int_extn...

Hi. I'm getting a strange sip issue, with latest cvs. I was tring the *8 extension for call pickup on sip, but I forget to define the callgroup & pickupgroup in sip.conf . Now when I dial *8 from the crisco phone and hangup, the channel in asterisk don't go down and I'm not able to dial from the phone again. If I do a softhangup on the rem. console it does nothing and the console simply stop responds. I can type commands but don't get anything back. I must quit the console and get back t...

Looking for a router to connect to a 5/50 cable modem that works with Sip. A Crisco RVS4000 installed now has real problems with Sip, one-way audio and throughput not up to the WAN speed. No VPN needed, something affordable, $200-$350 US range. Every thing I looked at in that range had some reported problem except pfSense in a ATX box. Any recommendations or comments apprecia...

An interesting article for those needing ammunition to sell Asterisk within their organisation or to others: "Is open source IP telephony ready for prime time? Yes" by Zenas Hutcheson, St. Paul Venture Capital Network World, 06/07/04 http://www.nwfusion.com/columnists/2004/0607faceoffyes.html On a related note, they also have an article arguing the contrary position (see link within

...phone is ringing, I can pickup the call from another sip phone, but the first one keeps playing a loud busy signal... that don't go away until I receive another call or go off hook and then on hook on the first phone. I think that could be a budgetone bug on BYE command, since the snom and the crisco works ok... But anyway I attached the log file (233 is the called, 225 is the one who pickups via *8). Anyone experienced that? Matteo. -------------- next part -------------- asterisk*CLI> sip debug SIP Debugging Enabled -- Accepting unauthenticated call from 213.140.14.155, requested fo...