Eric Wieling
2003-Feb-16 14:00 UTC
[Asterisk-Users] IConnectHere: Outbound working, Inbound not working....
Thanks to a response to my earlier post I have outbound calling working from a SIP Softphone to numbers on the PSTN via iconnecthere. Now I'm trying to get inbound calling working. It seems that when a call comes into Asterisk via iconnecthere it always falls into the "s" extention. I have two phone numbers with iconnecthere and want to route calls to each of the numbers differently. Can this be done? If so, how? Basically I want callers to 1813342XXXX to be routed to 2102 and calls to 1202454XXXX to be routed to 2103 Thanks in advance for any help! Here are my config files: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls register=1813342XXXX:XXXXXX at sipauth.deltathree.com register=1202454XXXX:XXXXXX at sipauth.deltathree.com [iconnecthere] type=friend username=XXXXXX secret=XXXXXX host=sipauth.deltathree.com context=default [2102] type=friend secret=XXXXXX host=dynamic context=default [2103] type=friend secret=XXXXXX host=dynamic context=default ; ; Extention Configuration for Asterisk ; [default] exten => 2102,1,Dial,SIP/2102 at 2102 exten => 2102,2,Voicemail(2102) exten => 2103,1,Dial,SIP/2103 at 2103 exten => 2103,2,Voicemail(2103) exten => _91XXXXXXXXXX,1,Dial,SIP/${EXTEN-1}@iconnecthere exten => _91XXXXXXXXXX,2,Congestion exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer ; Answer the line exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten => s,5,Directory,default exten => t,1,Goto(#,1) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again"
Mark Spencer
2003-Feb-16 14:58 UTC
[Asterisk-Users] IConnectHere: Outbound working, Inbound not working....
> Basically I want callers to 1813342XXXX to be routed to 2102 and > calls to 1202454XXXX to be routed to 2103 > > Thanks in advance for any help!I've added something (untested) to CVS which will allow you to do something like: register => username:password at siphost.com/localexten That way the call should be directed to the specified local extension. If it doesn't work, find me on IRC and I'll help you with it. Mark
Oliver Brandt
2003-Feb-17 13:15 UTC
[Asterisk-Users] IConnectHere: Outbound working, Inbound not working....
Hi,> Now I'm trying to get inbound calling working. It seems that > when a call comes into Asterisk via iconnecthere it always falls > into the "s" extention. I have two phone numbers with > iconnecthere and want to route calls to each of the numbers > differently. Can this be done? If so, how? > > Basically I want callers to 1813342XXXX to be routed to 2102 and > calls to 1202454XXXX to be routed to 2103I have not worke with incomming calls through iconnect so I'm not shure if this is what you need, but how about setting different contexts for your two iconnect numbers in your sip.conf? Then each should drop into its own context in the extensions.conf and you can do what ever you want with the calls. Hope this heped! Good luck! Oliver> > Thanks in advance for any help! > > Here are my config files: > > ; > ; SIP Configuration for Asterisk > ; > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0 ; Address to bind to > context = default ; Default for incoming calls > register=1813342XXXX:XXXXXX at sipauth.deltathree.com > register=1202454XXXX:XXXXXX at sipauth.deltathree.com > > [iconnecthere] > type=friend > username=XXXXXX > secret=XXXXXX > host=sipauth.deltathree.com > context=default > > [2102] > type=friend > secret=XXXXXX > host=dynamic > context=default > > [2103] > type=friend > secret=XXXXXX > host=dynamic > context=default > > ; > ; Extention Configuration for Asterisk > ; > [default] > > exten => 2102,1,Dial,SIP/2102 at 2102 > exten => 2102,2,Voicemail(2102) > > exten => 2103,1,Dial,SIP/2103 at 2103 > exten => 2103,2,Voicemail(2103) > > exten => _91XXXXXXXXXX,1,Dial,SIP/${EXTEN-1}@iconnecthere > exten => _91XXXXXXXXXX,2,Congestion > > exten => s,1,Wait,1 ; Wait a second, just for fun > exten => s,2,Answer ; Answer the line > exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds > exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds > exten => s,5,Directory,default > > exten => t,1,Goto(#,1) ; If they take too long, give up > exten => i,1,Playback(invalid) ; "That's not valid, try again" > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users