| Rev | Author | # | Line |
|---|---|---|---|
| 5 | TonyAllen | 1 | An Example setup for connecting from Asterisk to sipphone.com. |
| 2 | |||
| 3 | Note: If you are behind a router/firewall, you'll want to forward the ports 5004, 5060, and 69 to your local ip with the asterisk server on it. | ||
| 4 | |||
| 5 | !!extensions.conf | ||
| 6 | |||
| 7 | <verbatim> | ||
| 8 | [globals] | ||
| 9 | . | ||
| 10 | . | ||
| 11 | SIPPHONEUSERID=1747xxxXXXX | ||
| 12 | MYNAME=Bob | ||
| 13 | [[dialout] | ||
| 14 | . | ||
| 15 | . | ||
| 16 | include => sip-forced | ||
| 17 | include => from-sipphone | ||
| 18 | |||
| 19 | |||
| 20 | ; Check to see if the called number starts with a "6" and | ||
| 21 | ; if so, set the call parameters and bounce the call to the | ||
| 22 | ; SipPhone.com SIP server. | ||
| 23 | ; | ||
| 24 | ; NOTE: Calls to unknown users will result in "invalid extension" | ||
| 25 | ; message being played. | ||
| 26 | ; | ||
| 27 | [[sip-forced] | ||
| 28 | |||
| 29 | exten => _6.,1,SetCallerID(${SIPPHONEUSERID}) | ||
| 30 | exten => _6.,2,SetCIDName(${MYNAME}) | ||
| 31 | exten => _6.,3,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com) | ||
| 32 | exten => _6.,4,Playback(invalid) | ||
| 33 | exten => _6.,5,Hangup | ||
| 34 | |||
| 35 | |||
| 36 | ; To receive calls inbound from SipPhone.com, we set the extension | ||
| 37 | ; to our SipPhone user id, in this case from the SIPPHONEUSERID variable | ||
| 38 | ; Changing the "Dial" | ||
| 39 | ; directive to something like this: | ||
| 40 | ; Dial(${PHONES1}&${PHONES2},15,Ttm) | ||
| 41 | ; would cause both lines to ring | ||
| 42 | ; | ||
| 43 | |||
| 44 | [from-sipphone] | ||
| 45 | exten => ${SIPPHONEUSERID},1,Dial(${PHONES1},30,Ttm) | ||
| 46 | exten => ${SIPPHONEUSERID},2,Voicemail2(u${PHONES1VM}) | ||
| 47 | exten => ${SIPPHONEUSERID},3,Hangup | ||
| 48 | </verbatim> | ||
| 49 | |||
| 50 | !!sip.conf | ||
| 51 | |||
| 52 | <verbatim> | ||
| 53 | [[general] | ||
| 54 | . | ||
| 55 | . | ||
| 56 | . | ||
| 57 | register=<YourSIPPhoneID>:<YourSIPPhonePassword>@proxy01.sipphone.com/<extension> | ||
| 58 | . | ||
| 59 | . | ||
| 60 | ;be sure to set your external ip if you're behind a router. | ||
| 61 | ;Especially if you want to use sipphone minutes | ||
| 62 | externip=xxx.xxx.xxx.xxx | ||
| 63 | . | ||
| 64 | . | ||
| 65 | ;uncomment this line as it should be in your default sip.conf | ||
| 66 | localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks | ||
| 6 | JohnMcPherson | 67 | |
| 5 | TonyAllen | 68 | |
| 69 | |||
| 70 | ;Note: the name is [proxy01.sipphone.com] so that incoming calls work correctly | ||
| 71 | [proxy01.sipphone.com] | ||
| 72 | type=friend | ||
| 73 | secret=<YourSIPPhonePassword> | ||
| 74 | username=<YourSIPPhoneID> | ||
| 75 | host=proxy01.sipphone.com | ||
| 76 | dtmfmode=inband | ||
| 77 | context=dialout ;or your context which includes the dialing rules | ||
| 78 | nat=yes | ||
| 79 | qualify=no | ||
| 80 | reinvite=no | ||
| 81 | canreinvite=no | ||
| 7 | AdamWentworth | 82 | fromdomain=proxy01.sipphone.com ; I was unable to dial out to PSTN using SIP Minutes without this line, Failed INVITE :) |
| 5 | TonyAllen | 83 | </verbatim> |
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