SipDevicesAsterisk » History » Version 1
Anonymous, 11/14/2010 11:22 AM
| 1 | 1 | == Asterisk PBX == |
|
|---|---|---|---|
| 2 | |||
| 3 | The SIP2SIP platform consists of several servers, addressed by DNS SRV records. Asterisk, however is currently unable to handle more that one result for a DNS SRV lookup, so the configuration needed for getting it work with the SIP2SIP service is a bit confusing. This wiki page helps clarify that hopefully. |
||
| 4 | |||
| 5 | === Versions 1.4 and 1.6.x === |
||
| 6 | |||
| 7 | '''dnsmgr.conf''' |
||
| 8 | |||
| 9 | {{{ |
||
| 10 | [general] |
||
| 11 | enable=yes |
||
| 12 | }}} |
||
| 13 | |||
| 14 | |||
| 15 | '''sip.conf''' |
||
| 16 | |||
| 17 | {{{ |
||
| 18 | [general] |
||
| 19 | ... |
||
| 20 | srvlookup=yes |
||
| 21 | ... |
||
| 22 | |||
| 23 | register => 2233XXXXX:password@sip2sip.info/2233XXXXX |
||
| 24 | ... |
||
| 25 | |||
| 26 | [authentication] |
||
| 27 | |||
| 28 | [sip2sip](!) |
||
| 29 | type=peer |
||
| 30 | canreinvite=no |
||
| 31 | nat=yes |
||
| 32 | qualify=yes |
||
| 33 | domain=sip2sip.info |
||
| 34 | fromdomain=sip2sip.info |
||
| 35 | outboundproxy=proxy.sipthor.net |
||
| 36 | fromuser=2233XXXXX |
||
| 37 | username=2233XXXXX |
||
| 38 | secret=password |
||
| 39 | insecure=invite |
||
| 40 | context=from-sip2sip |
||
| 41 | |||
| 42 | [sip2sip-0](sip2sip) |
||
| 43 | host=sip2sip.info |
||
| 44 | |||
| 45 | [sip2sip-1](sip2sip) |
||
| 46 | host=81.23.228.129 |
||
| 47 | |||
| 48 | [sip2sip-2](sip2sip) |
||
| 49 | host=81.23.228.150 |
||
| 50 | |||
| 51 | [sip2sip-3](sip2sip) |
||
| 52 | host=85.17.186.7 |
||
| 53 | }}} |
||
| 54 | |||
| 55 | |||
| 56 | '''extensions.conf''' |
||
| 57 | {{{ |
||
| 58 | [from-users] |
||
| 59 | ; Dialing the SIP2SIP echo test |
||
| 60 | ; IMPORTANT: all outbound calls to SIP2SIP need to be done using the 'sip2sip-0' peer |
||
| 61 | exten => 1234,1,Dial(SIP/3333@sip2sip-0) |
||
| 62 | |||
| 63 | [from-sip2sip] |
||
| 64 | ; 2233XXXXX is your SIP2SIP username, NOT a dialplan pattern |
||
| 65 | exten => 2233XXXXX,1,NoOp(--Incoming call from ${CALLERID(all)}) |
||
| 66 | exten => 2233XXXXX,n,Dial(SIP/phone1, 60) |
||
| 67 | }}} |
||
| 68 | |||
| 69 | |||
| 70 | === Version 1.8 === |
||
| 71 | |||
| 72 | '''dnsmgr.conf''' |
||
| 73 | |||
| 74 | {{{ |
||
| 75 | [general] |
||
| 76 | enable=yes |
||
| 77 | }}} |
||
| 78 | |||
| 79 | |||
| 80 | '''sip.conf''' |
||
| 81 | |||
| 82 | {{{ |
||
| 83 | [general] |
||
| 84 | ... |
||
| 85 | srvlookup=yes |
||
| 86 | ... |
||
| 87 | |||
| 88 | register => 2233XXXXX:password@sip2sip.info/2233XXXXX |
||
| 89 | ... |
||
| 90 | |||
| 91 | [authentication] |
||
| 92 | |||
| 93 | [sip2sip](!) |
||
| 94 | type=peer |
||
| 95 | canreinvite=no |
||
| 96 | nat=yes |
||
| 97 | qualify=yes |
||
| 98 | domain=sip2sip.info |
||
| 99 | fromdomain=sip2sip.info |
||
| 100 | outboundproxy=proxy.sipthor.net |
||
| 101 | fromuser=2233XXXXX |
||
| 102 | defaultuser=2233XXXXX |
||
| 103 | secret=password |
||
| 104 | insecure=invite |
||
| 105 | context=from-sip2sip |
||
| 106 | |||
| 107 | [sip2sip-0](sip2sip) |
||
| 108 | host=sip2sip.info |
||
| 109 | |||
| 110 | [sip2sip-1](sip2sip) |
||
| 111 | host=81.23.228.129 |
||
| 112 | |||
| 113 | [sip2sip-2](sip2sip) |
||
| 114 | host=81.23.228.150 |
||
| 115 | |||
| 116 | [sip2sip-3](sip2sip) |
||
| 117 | host=85.17.186.7 |
||
| 118 | }}} |
||
| 119 | |||
| 120 | |||
| 121 | '''extensions.conf''' |
||
| 122 | {{{ |
||
| 123 | [from-users] |
||
| 124 | ; Dialing the SIP2SIP echo test |
||
| 125 | ; IMPORTANT: all outbound calls to SIP2SIP need to be done using the 'sip2sip-0' peer |
||
| 126 | exten => 1234,1,Dial(SIP/3333@sip2sip-0) |
||
| 127 | |||
| 128 | [from-sip2sip] |
||
| 129 | ; 2233XXXXX is your SIP2SIP username, NOT a dialplan pattern |
||
| 130 | exten => 2233XXXXX,1,NoOp(--Incoming call from ${CALLERID(all)}) |
||
| 131 | same => n,Dial(SIP/phone1, 60) |
||
| 132 | }}} |