bubblestar

sip.conf settings for Asterisk 1.8

1. [general]
   
2. ; Global Settings
   
3. bindport = 5060                                        ; Port to bind to (SIP is 5060)
   
4. bindaddr = 0.0.0.0                                        ; Bind all addresses on machine
   
5. realm = asterisk
   
6. useragent = myuseragent
   
7. ;sdpsession = myuseragent
   
8. allowguest = yes                                        ; Allow or reject guest calls ; set allowguest  =  no for security reason                         
   
9. allowsubscribe = yes
   
10. canreinvite = no
    
11. insecure = port,invite
    
12. srvlookup = yes
    
13. ;qualifyfreq = 60
    
14. ;qualifygap = 100
    
15. ;qualifypeers = 1
    
16. callevents = no
    
17. ;allowexternalinvites = yes
    
18. allowexternaldomains = yes
    
19. alwaysauthreject = yes                        
    
20. allowoverlap = no                                        ; Disable overlap dialing support (Default is yes)                     
    
21. allowtransfer = yes                                        ; Disable all transfers                        
    
22. videosupport = no
    
23. callcounter = yes
    
24. t38pt_udptl = yes,fec,maxdatagram = 400
    
25. faxdetect = yes
    
26. 
    
27. 
    
28. ; Network QoS Settings
    
29. tos_sip = CS3                                        ; Sets TOS for SIP packets                    
    
30. tos_audio = ef                                        ; Sets TOS for RTP audio packets.                             
    
31. tos_video = AF41                                        ; Sets TOS for RTP video packets                                
    
32. cos_sip = 3
    
33. cos_audio = 5
    
34. cos_video = 4
    
35. cos_text = 3
    
36. jbenable = no
    
37. jbforce = no
    
38. 
    
39. ; Network Settings
    
40. externrefresh = 10
    
41. externhost = your_ddns_name                                ; DDNS
    
42. fromdomain = your_ddns_name                        ; Optional - force a particular domain        
    
43. localnet = xxx.xxx.xxx.xxx/255.255.255.0                        ; Asterisk network address and mask           
    
44. stunaddr =
    
45. autodomain = no
    
46. 
    
47. ; Global Signaling Settings
    
48. disallow = all
    
49. allow = ulaw
    
50. allow = alaw
    
51. allow = gsm                                                ; GSM needs low bandwidth than ulaw and alaw
    
52. allow = g729
    
53. allow = slin
    
54. faxdetect = on
    
55. rtptimeout = 60
    
56. rtpholdtimeout = 300
    
57. rtpkeepalive = 20                                               ; Send a keepalive ever 20 Seconds if using NAT
    
58. maxexpiry = 3600                                        ; **Engin & BBP Global this if necessary
    
59. minexpiry = 60                                       
    
60. defaultexpiry = 240                                        ; **Engin users: include users: include this if necessary
    
61. registerattempts = 0
    
62. registertimeout = 20
    
63. relaxdtmf = yes
    
64. notifyringing = yes
    
65. notifyhold = yes
    
66. notifycid = yes
    
67. pedantic = no
    
68. progressinband = never
    
69. promiscredir = no
    
70. 
    
71. ; Default Settings
    
72. nat = yes
    
73. dtmfmode = rfc2833
    
74. qualify = yes
    
75. context = default                                        ; Send unknown SIP incoming callers to this context
    
76. language = en
    
77. musicclass = default
    
78. mohinterpret = default
    
79. mohsuggest = default
    
80. 
    
81. 
    
82. [authentication]
    
83. 
    
84. 
    
85. 
    
86. [my-settings](!)                                        ; template for the phones
    
87. type = friend                                        ; is both peer (out) and user (in)
    
88. qualify = yes
    
89. nat = yes
    
90. host = dynamic
    
91. dtmfmode = auto
    
92. allow = ulaw,alaw,gsm,g729,slin
    
93. context = yourcontextname
    
94. canreinvite = no                                        ; set "canreinvite = yes" for all internal extensions. This will let RTP directly flow from Line 1 or other ATA to PSTN without going through Asterisk, thus to minimize the delay
    
95. insecure = port,invite
    
96. port = 5060
    
97. ;musiconhold = default
    
98. ;musciclass = default
    
99. ;deny=0.0.0.0/0.0.0.0
    
100. ;permit=xxx.xxx.xxx/255.255.255.0
     
101. 
     
102. 
     
103. [6001](my-settings)
     
104. defaultuser = 6001
     
105. secret = very_secret_code
     
106. mailbox = 6001@default
     
107. vmsecret = 6001
     
108. dial = SIP/6001                                                       
     
109. callerid = "who_is_who" <>
     
110. ;accountcode =        
     
111. ;callgroup = 1,3-4                                        ; members of groups 1,3 to 4
     
112. ;pickupgroup = 1,2-4                                        ; member of "pickup" groups 1,2 to 4
     
113. call-limit = 10
     
114. musiconhold = default
     
115. musciclass = default
     
116. 
     
117. 
     
118. [6002](phone-settings)
     
119. defaultuser = 6002
     
120. secret = very_secret_code
     
121. mailbox = 6002@default
     
122. vmsecret = 6002
     
123. dial = SIP/6002
     
124. callerid = "who_am_i" <>
     
125. ;accountcode =                        
     
126. ;callgroup = 1,3-4                                        ; members of groups 1,3 to 4       
     
127. ;pickupgroup = 1,2-4                                        ; member of "pickup" groups 1,2 to 4
     
128. call-limit = 10
     
129. musiconhold = friends
     
130. musciclass = friends

複製代碼

bubblestar

No particular settings required for extensions.conf.  Just config as you do in Asterisk 1.4 or 1.6.

Some terms or names for creation of extension line in the file of sip.conf are different.  So, please pay special care for such changes.  For instance, user name is changed to defaultuser.  Insecure is now set as "port,invite" instead of "very" as that in Asterisk 1.4

角色

After having tried many combinations, finally I got the simplest form in sip.conf in order to get one extension registered as follows:

sip.conf

1. [2000]
   
2. type = friend
   
3. secret=2000
   
4. qualify=yes
   
5. host=dynamic
   
6. context=internal

複製代碼

bubblestar

簡單是美，高興聽到你能做到。

有關你原本不能成功的設定，可否貼上來讓大家分享一下原因呢? Thanks

bubblestar

回復 18# 角色


   
另外，你可以慢慢把insecure、qualify、 allow 之類的東西逐一加上去試試，那樣或者可以知道之前不能成功註冊的原因了。至樣最重要的 nat = yes 是不能缺少的了，你能否通過router跟外界順利通話，這個也是關鍵。

ckleea

Bubblestar Ching is particularly in doing various fine settings. I am a pioneer to test.

ckleea

I am also in flavour to use IAX connection. Zoiper is my best friend

角色

When I have time, I shall add more switches on it.

YH

角色

There is another example to demonstrate the use of the sip.conf to set up the Asterisk server as follows:

1. [2000]
   
2. type = friend
   
3. secret=2000
   
4. nat=yes
   
5. qualify=yes
   
6. host=dynamic
   
7. context=internal
   
8. 
   
9. [2001]
   
10. type = friend
    
11. secret=2001
    
12. nat=yes
    
13. qualify=yes
    
14. host=dynamic
    
15. context=internal

複製代碼

Please note the parameter "nat", if nat=no, Zoiper is not got registrated; and if nat=yes, Zoiper is got registrated. Siemens DECT IP Phone can register regardless of the settings of the parameter "nat".

Therefore, it is better to set the parameter "nat=yes" would be better.

YH

角色

I know the reason that I am not able to get registration. There is a type for the type, I typed it as typp rather the correct one type.

Once the word "typp" is replaced by "type", everything are working fine.

YH

ckleea

回復 25# 角色

you are typing too fast with vi

角色

Maybe.

YH

bubblestar

回復 21# ckleea


    I am flattered.  I think all of us are aiming at the same goal but just from different approach.

bubblestar

回復 25# 角色


    Congratulations! Welcome on board the Asterisk 1.8 and we are on the same track now.

角色

Yeah!

角色