search for: huntstop

...s] OUTBOUNDTRUNK=SIP/Cisco1760 [outbound-local] exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9NXXXXXX,n,Congestion() exten => _9NXXXXXX,n,Hangup() -----------Cisco 1760 config ---------- dial-peer voice 100 pots (This line that is set to preference 2 does not work) huntstop preference 2 destination-pattern .T port 0/0 forward-digits all ! dial-peer voice 2212 pots (This line that is set to Preference 1 is the one that works) huntstop preference 1 destination-pattern .T port 0/1 forward-digits all Thanks, Jimmy

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

...2 pots description Otro catch all, esta vez para FXS incoming called-number T ! dial-peer voice 2700 voip destination-pattern T session target sip-server ! dial-peer voice 1 voip voice-class codec 1 incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 102700 pots huntstop destination-pattern 2700 port 2/0 ! sip-ua retry register 5 registrar ipv4:192.168.0.200 expires 3600 ! ! line con 0 login local transport preferred all transport output all line aux 0 transport preferred all transport output all line vty 0 4 password asterisk26 login transport preferr...

I see a lot of comments but no actual show runs. Can someone post a 7206 config. I am having a dickens of a time getting calls to pass. I currently have the following loaded. Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6, RELEASE SOFTWARE (fc2) Thanks !!! Jerry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 /

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk