SipDevicesAsterisk
Version 3 (Adrian Georgescu, 10/15/2011 05:15 pm)
| 1 | 1 | == Asterisk PBX == |
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|---|---|---|---|
| 2 | 1 | ||
| 3 | 3 | Adrian Georgescu | [[TOC(SipDevices*, depth=2)]] |
| 4 | 3 | Adrian Georgescu | |
| 5 | 2 | Adrian Georgescu | SIP2SIP service consists of several OpenSIPS servers running in difference data centers using SIP Thor distributed architecture, which is using DNS SRV records to point to the actual servers. Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it work with the SIP2SIP service is not trivial. This wiki page helps clarify it. |
| 6 | 1 | ||
| 7 | 1 | === Versions 1.4 and 1.6.x === |
|
| 8 | 1 | ||
| 9 | 1 | '''dnsmgr.conf''' |
|
| 10 | 1 | ||
| 11 | 1 | {{{ |
|
| 12 | 1 | [general] |
|
| 13 | 1 | enable=yes |
|
| 14 | 1 | }}} |
|
| 15 | 1 | ||
| 16 | 1 | ||
| 17 | 1 | '''sip.conf''' |
|
| 18 | 1 | ||
| 19 | 1 | {{{ |
|
| 20 | 1 | [general] |
|
| 21 | 1 | ... |
|
| 22 | 1 | srvlookup=yes |
|
| 23 | 1 | ... |
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| 24 | 1 | ||
| 25 | 1 | register => 2233XXXXX:password@sip2sip.info/2233XXXXX |
|
| 26 | 1 | ... |
|
| 27 | 1 | ||
| 28 | 1 | [authentication] |
|
| 29 | 1 | ||
| 30 | 1 | [sip2sip](!) |
|
| 31 | 1 | type=peer |
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| 32 | 1 | canreinvite=no |
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| 33 | 1 | nat=yes |
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| 34 | 1 | qualify=yes |
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| 35 | 1 | domain=sip2sip.info |
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| 36 | 1 | fromdomain=sip2sip.info |
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| 37 | 1 | outboundproxy=proxy.sipthor.net |
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| 38 | 1 | fromuser=2233XXXXX |
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| 39 | 1 | username=2233XXXXX |
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| 40 | 1 | secret=password |
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| 41 | 1 | insecure=invite |
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| 42 | 1 | context=from-sip2sip |
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| 43 | 1 | ||
| 44 | 1 | [sip2sip-0](sip2sip) |
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| 45 | 1 | host=sip2sip.info |
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| 46 | 1 | ||
| 47 | 1 | [sip2sip-1](sip2sip) |
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| 48 | 1 | host=81.23.228.129 |
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| 49 | 1 | ||
| 50 | 1 | [sip2sip-2](sip2sip) |
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| 51 | 1 | host=81.23.228.150 |
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| 52 | 1 | ||
| 53 | 1 | [sip2sip-3](sip2sip) |
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| 54 | 1 | host=85.17.186.7 |
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| 55 | 1 | }}} |
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| 56 | 1 | ||
| 57 | 1 | ||
| 58 | 1 | '''extensions.conf''' |
|
| 59 | 1 | {{{ |
|
| 60 | 1 | [from-users] |
|
| 61 | 1 | ; Dialing the SIP2SIP echo test |
|
| 62 | 1 | ; IMPORTANT: all outbound calls to SIP2SIP need to be done using the 'sip2sip-0' peer |
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| 63 | 1 | exten => 1234,1,Dial(SIP/3333@sip2sip-0) |
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| 64 | 1 | ||
| 65 | 1 | [from-sip2sip] |
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| 66 | 1 | ; 2233XXXXX is your SIP2SIP username, NOT a dialplan pattern |
|
| 67 | 1 | exten => 2233XXXXX,1,NoOp(--Incoming call from ${CALLERID(all)}) |
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| 68 | 1 | exten => 2233XXXXX,n,Dial(SIP/phone1, 60) |
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| 69 | 1 | }}} |
|
| 70 | 1 | ||
| 71 | 1 | ||
| 72 | 1 | === Version 1.8 === |
|
| 73 | 1 | ||
| 74 | 1 | '''dnsmgr.conf''' |
|
| 75 | 1 | ||
| 76 | 1 | {{{ |
|
| 77 | 1 | [general] |
|
| 78 | 1 | enable=yes |
|
| 79 | 1 | }}} |
|
| 80 | 1 | ||
| 81 | 1 | ||
| 82 | 1 | '''sip.conf''' |
|
| 83 | 1 | ||
| 84 | 1 | {{{ |
|
| 85 | 1 | [general] |
|
| 86 | 1 | ... |
|
| 87 | 1 | srvlookup=yes |
|
| 88 | 1 | ... |
|
| 89 | 1 | ||
| 90 | 1 | register => 2233XXXXX:password@sip2sip.info/2233XXXXX |
|
| 91 | 1 | ... |
|
| 92 | 1 | ||
| 93 | 1 | [authentication] |
|
| 94 | 1 | ||
| 95 | 1 | [sip2sip](!) |
|
| 96 | 1 | type=peer |
|
| 97 | 1 | canreinvite=no |
|
| 98 | 1 | nat=yes |
|
| 99 | 1 | qualify=yes |
|
| 100 | 1 | domain=sip2sip.info |
|
| 101 | 1 | fromdomain=sip2sip.info |
|
| 102 | 1 | outboundproxy=proxy.sipthor.net |
|
| 103 | 1 | fromuser=2233XXXXX |
|
| 104 | 1 | defaultuser=2233XXXXX |
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| 105 | 1 | secret=password |
|
| 106 | 1 | insecure=invite |
|
| 107 | 1 | context=from-sip2sip |
|
| 108 | 1 | ||
| 109 | 1 | [sip2sip-0](sip2sip) |
|
| 110 | 1 | host=sip2sip.info |
|
| 111 | 1 | ||
| 112 | 1 | [sip2sip-1](sip2sip) |
|
| 113 | 1 | host=81.23.228.129 |
|
| 114 | 1 | ||
| 115 | 1 | [sip2sip-2](sip2sip) |
|
| 116 | 1 | host=81.23.228.150 |
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| 117 | 1 | ||
| 118 | 1 | [sip2sip-3](sip2sip) |
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| 119 | 1 | host=85.17.186.7 |
|
| 120 | 1 | }}} |
|
| 121 | 1 | ||
| 122 | 1 | ||
| 123 | 1 | '''extensions.conf''' |
|
| 124 | 1 | {{{ |
|
| 125 | 1 | [from-users] |
|
| 126 | 1 | ; Dialing the SIP2SIP echo test |
|
| 127 | 1 | ; IMPORTANT: all outbound calls to SIP2SIP need to be done using the 'sip2sip-0' peer |
|
| 128 | 1 | exten => 1234,1,Dial(SIP/3333@sip2sip-0) |
|
| 129 | 1 | ||
| 130 | 1 | [from-sip2sip] |
|
| 131 | 1 | ; 2233XXXXX is your SIP2SIP username, NOT a dialplan pattern |
|
| 132 | 1 | exten => 2233XXXXX,1,NoOp(--Incoming call from ${CALLERID(all)}) |
|
| 133 | 1 | same => n,Dial(SIP/phone1, 60) |
|
| 134 | 1 | }}} |