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