角色

怎样show advanced options?

1. 点击在右上角的button

下載 (13.61 KB)

2013-10-14 20:34

2. 留意最底部的变化

下載 (14.99 KB)

2013-10-14 20:34

3. 多处的options

下載 (3.55 KB)

2013-10-14 20:34

角色

怎样更改login name and password?

有两个方法去修改login name and password.那些credentials放在manager.conf。

第一种方法
ssh login + vi /etc/asterisk/manager.conf

At the end of manager.conf

1. [admin]
   
2. secret=admin
   
3.   read=system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
   
4.   write=system,call,agent,user,config,command,reporting,originate

複製代碼

在上面"[admin]",是login name：admin。Password放在secret，那么password就是admin。你可以作适当的修改就可以。

第二种方法
1. 先enable advanced options Link
2. 在左右边的advanced otpions里，点击“File Editor”，然后选manager.conf

下載 (4.53 KB)

2013-10-14 21:51

修改secret的content就可以更改password

下載 (8.37 KB)

2013-10-14 21:51

修改login name就比较复杂一点。就先先creat 一个新的context，然后copy之前”-admin“下的资料，然后save，再apply changes。

carw318

在 ebay 找到這model 喎 "SIEMENS GIGASET VoIP & FIXED LINE CORDLESS PHONE C610A-IP" USD99.99 + USD25.08(shipping fee) , 好似不錯咁 ~~

http://www.ebay.com/itm/SIEMENS- ... 1e8045c0bb#shpCntId

角色

我们一般都用三个handsets, 看来不出来eBay这个offer有几个handsets?

角色

Internal SIP call

Extension 6000 = SIP Client 1(IP Phone)
Extension 6001 = SIP Client 2(Zoiper softphone)

1. Create a dialplan

下載 (14.71 KB)

2013-10-15 23:16

下載 (12.53 KB)

2013-10-15 23:16

下載 (16.04 KB)

2013-10-15 23:16

2. Create two extensions

下載 (16.05 KB)

2013-10-15 23:16

2a. Create Extension 6000

下載 (28.17 KB)

2013-10-15 23:16

2b. Create Extension 6001

下載 (27.97 KB)

2013-10-15 23:16

3. Apply changes

下載 (8.24 KB)

2013-10-15 23:16

4. SIP Client 1 (IP Phone)

下載 (22.73 KB)

2013-10-15 23:16

5. SIP Client 2 (Zoiper Softphone)

下載 (21.75 KB)

2013-10-15 23:16

6. Test

下載 (36.25 KB)

2013-10-15 23:16

角色

Asterisk-GUI的Voicemail又出问题！

carw318

是什麼問題 ? 明天星期六 , 又開始我要set asterisk 了 , 要試下 setup trunks 先 !

角色

一set就出问题，连互打都不成功！说有error。因为我没有用过，估计是不会set。

角色

Echo Test

大家可以参考下面文章：

http://www.telecom-cafe.com/foru ... =4714&pid=24658

但是RP的Asterisk-GUi已经没有Custom这个application了。

角色

Asterisk-GUI Settings for NWT Internet/Broadband Phone

Please note that the softswitch (Huawei SoftX3000 V300R010) of NWT is the same as ComNet Phone using, therefore the settings for ComNet Phone is also applicable to NWT Phone.

1. <--- SIP read from UDP:203.176.254.198:5060 --->
   
2. INVITE sip:33445566@192.168.88.100:5060;user=phone SIP/2.0
   
3. Via: SIP/2.0/UDP 203.176.254.198:5060;branch=z9hG4bKblxcvluokkhnjxnjo78civlhv
   
4. Call-ID: SBCqlhluppq5cuxwcpxgmgcwulbygvvvgcy@SoftX3000
   
5. From: <sip:96331111@203.176.254.198>;tag=wq8xxuul-CC-22
   
6. To: <sip:33445566@58.153.143.183:30628;user=phone>
   
7. CSeq: 1 INVITE
   
8. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
   
9. Max-Forwards: 70
   
10. Supported: 100rel
    
11. User-Agent: Huawei SoftX3000 V300R010
    
12. Contact: <sip:96331111@203.176.254.198:5060;user=phone>
    
13. Content-Length: 427
    
14. Content-Type: application/sdp
    
15. 
    
16. v=0
    
17. o=HuaweiSoftX3000 93845972 93845972 IN IP4 203.176.254.198
    
18. s=Sip Call
    
19. c=IN IP4 203.176.254.198
    
20. t=0 0
    
21. m=audio 23684 RTP/AVP 8 0 18 4 2 97 98 101
    
22. a=rtpmap:8 PCMA/8000
    
23. a=rtpmap:0 PCMU/8000
    
24. a=rtpmap:18 G729/8000
    
25. a=rtpmap:4 G723/8000
    
26. a=rtpmap:2 G726-32/8000
    
27. a=rtpmap:97 G726-40/8000
    
28. a=rtpmap:98 G726-32/8000
    
29. a=rtpmap:101 telephone-event/8000
    
30. a=ptime:20
    
31. a=fmtp:101 0-15
    
32. a=fmtp:18 annexb=yes
    
33. a=fmtp:4 annexa=yes

複製代碼

1. Create the NWT VoIP Trunk

下載 (9.6 KB)

2013-11-7 00:24

and then re-edit again to add the follwing settings:

下載 (14.06 KB)

2013-11-7 00:24

2. Create the outgoing rule for NWT Trunk

下載 (16.92 KB)

2013-11-7 00:24

3. Create the dialplan to include the NWT VoIP Trunk

下載 (11.3 KB)

2013-11-7 00:24

4. Create a user to use the dialplan

下載 (35.75 KB)

2013-11-7 00:24

5. Inbound call settings

下載 (7.93 KB)

2013-11-7 00:24

6. Other settings for outbound call (if not set, callee will not able to receive the call)

下載 (12.03 KB)

2013-11-7 00:24

7. Apply changes

下載 (6.04 KB)

2013-11-7 00:24

8. Minimum 300 seconds

1. handle_response_register: Got 423 Interval too brief for service 333445566@ngn2.nwtbb.com, minimum is 300 seconds

複製代碼

Using the File Editor (Advanced mode) to add the switch "defaultexpirey=300" to the users.conf under the NWT trunk. For details, pleas take a look the following scripts:

下載 (10.24 KB)

2013-11-8 00:01

角色

VoIP Trunk除了NWT NetTalk，HKBN 2b，ComNet Phone，现在还可以用3G Modem！做Trunk。

SuiYan

係咩.
但係MANUAL人手SET trunk.

角色

Asterisk-GUI Settings for ComNet Phone

Please note the soft switch employed in ComNet Phone is a Huawei SoftSwitch "HuaweiSoftX3000"

1. <--- Transmitting (NAT) to 192.168.88.249:5060 --->
   
2. SIP/2.0 180 Ringing
   
3. Via: SIP/2.0/UDP 192.168.88.249:5060;branch=z9hG4bK-d8754z-b7b0e2881d6f8659-1---d8754z-;received=192.168.88.249;rport=5060
   
4. From: "6001"<sip:6001@192.168.88.100;transport=UDP>;tag=8a5ff34d
   
5. To: <sip:1878200@192.168.88.100;transport=UDP>;tag=as76214c7a
   
6. Call-ID: NDk3OGUwZTlmYjFiNTExNWQwZDJjMTRhM2U5MDdiOTM.
   
7. CSeq: 1 INVITE
   
8. Server: Asterisk PBX
   
9. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
   
10. Supported: replaces, timer
    
11. Contact: <sip:1878200@192.168.88.100:5060>
    
12. Content-Length: 0
    
13. 
    
14. 
    
15. <------------>
    
16.     -- SIP/trunk_1-00000025 is making progress passing it to SIP/6001-00000024
    
17. Audio is at 18618
    
18. Adding codec 100003 (ulaw) to SDP
    
19. 
    
20. <--- SIP read from UDP:202.0.179.3:5060 --->
    
21. SIP/2.0 200 OK
    
22. From: "asterisk" <sip:85235010617@192.168.88.100:5060>;tag=as29b58819
    
23. To: <sip:1878200@202.0.179.3>;tag=fa4e3fa8
    
24. CSeq: 103 INVITE
    
25. Call-ID: 2c672c914ce39bbb6f170df52d4e9103@192.168.88.100:5060
    
26. Via: SIP/2.0/UDP 192.168.88.100:5060;branch=z9hG4bK571a7528;rport=5060
    
27. Contact: <sip:1878200@202.0.179.3:5060;user=phone>
    
28. Content-Length: 200
    
29. Content-Type: application/sdp
    
30. 
    
31. v=0
    
32. o=HuaweiSoftX3000 7283948 7283949 IN IP4 202.0.179.3
    
33. s=Sip Call
    
34. c=IN IP4 202.0.179.3
    
35. t=0 0
    
36. m=audio 18220 RTP/AVP 0 97
    
37. a=rtpmap:0 PCMU/8000
    
38. a=rtpmap:97 telephone-event/8000
    
39. a=fmtp:97 0-15
    
40. <------------->
    
41. --- (9 headers 9 lines) ---
    
42. Found RTP audio format 0
    
43. Adding codec 100004 (alaw) to SDP
    
44. Adding non-codec 0x1 (telephone-event) to SDP

複製代碼

1. Go to the left-side menu, select Options and then followed by choosing the Show Advanced Options. Click the SIP Settings and tick the checkbox of "Pedantic", as shown in the following figure (without the tick at "Pedantic" check box, you are not able to get the registration worked)

下載 (17.93 KB)

2013-11-8 00:43

2. Create a VoIP SIP Trunk

下載 (9.76 KB)

2013-11-8 01:12

And then re-edit again to add more itemss

下載 (15.63 KB)

2013-11-8 01:12

3. Create a dialling prefix

下載 (16.91 KB)

2013-11-8 01:12

Edit dialplan to add outbound SIP Trunk "NWT NetTalk"

下載 (10.89 KB)

2013-11-8 01:12

Inbound settings

下載 (7.95 KB)

2013-11-8 01:12

Edit asterisk.conf and then add the term "internal_timing=yes"

下載 (13.93 KB)

2013-11-8 01:12

角色

Asterisk-GUI Settings for HKBN 2b Trunk

1. Create VoIP Trunk

下載 (9.74 KB)

2013-11-8 21:54

Further modify the trunk by editing.

下載 (15.47 KB)

2013-11-8 21:54

Using an editor like vi to add the following line at the end of /etc/hosts
203.80.89.135   s2hkbntel.net s21.hkbntel.net

2. Create dial prefix

下載 (16.8 KB)

2013-11-8 21:54

3. Add the default dial plan

下載 (9.81 KB)

2013-11-8 21:54

4. Add the incoming settings:

下載 (8.03 KB)

2013-11-8 21:54

5. Reboot the system

6. Check the system status

下載 (6.4 KB)

2013-11-8 21:59

magiccat

終於可以回覆了，在這我要強烈地多謝角色的幫助，才可成功在asterisk gui 使用nwt .
剛去完旅行，有時間會寫下我個人在synology compile asterisk 的流程， iptables fail2ban .