I have a setup with a SIP DID inbound, and several SIP phones inside. Obviously if the SIP phones are off/unplugged/otherwise not available, incoming calls ring busy. My extensions.conf looks like this for inbound calls: exten => _1xxxxxxxxxx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr) So what could I do to send the call to voicemail if none of the extensions are online? Thanks! jlc
Eric "ManxPower" Wieling
2008-Sep-08 20:03 UTC
[asterisk-users] SIP Extension Config Issue
You're joking, right? exten => _1xxxxxxxxxx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr) exten => _1xxxxxxxxxx,n,Voicemail(mailbox at vmcontext) Use whatever voice mailbox and voicemail context you want. Joseph L. Casale wrote:> I have a setup with a SIP DID inbound, and several SIP phones inside. > Obviously if the SIP phones are off/unplugged/otherwise not available, > incoming calls ring busy. My extensions.conf looks like this for inbound > calls: > > exten => _1xxxxxxxxxx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr) > > So what could I do to send the call to voicemail if none of the extensions > are online? > > Thanks! > jlc > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide.
>exten => _1xxxxxxxxxx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr) >exten => _1xxxxxxxxxx,n,Voicemail(mailbox at vmcontext) > >Use whatever voice mailbox and voicemail context you want.Well, its not advancing when *no* phones are online, just ringing busy. It does however step through just fine when they *are* online. I assumed that since it advances through correctly when they are online there is something else that happens to asterisk when no peers inside are registered. Thanks for the help :) jlc
Joseph L. Casale wrote:>> exten => _1xxxxxxxxxx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr) >> exten => _1xxxxxxxxxx,n,Voicemail(mailbox at vmcontext) >>exten => _1xxxxxxxxxx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr) exten => _1xxxxxxxxxx,n,NoOP(Dial Status: ${DIALSTATUS}) exten => _1xxxxxxxxxx,n,NoOP(Hangup Cause: ${HANGUPCAUSE}) exten => _1xxxxxxxxxx,n,Gosub(s-${DIALSTATUS},s,1) [s-BUSY] exten => s,1,Voicemail(${ARG1}@sip|b) Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."