【Raspberry Pi】——Asterisk-GUI + VoIP Trunks （NWT NetTalk, HKBN 2b, ComNet Phone） - Asterisk - VoIP 研究室 - 電訊茶室

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發表於 2013-10-25 00:21 | 顯示全部帖子

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。

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32^#

發表於 2013-11-7 00:28 | 顯示全部帖子

本帖最後由角色於 2013-11-8 00:01 編輯

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.

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

複製代碼

1. Create the NWT VoIP Trunk

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

2. Create the outgoing rule for NWT Trunk

3. Create the dialplan to include the NWT VoIP Trunk

4. Create a user to use the dialplan

5. Inbound call settings

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

7. Apply changes

8. Minimum 300 seconds

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:

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33^#

發表於 2013-11-7 22:38 | 顯示全部帖子

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

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34^#

發表於 2013-11-8 00:35 | 顯示全部帖子

本帖最後由角色於 2013-11-8 01:18 編輯

Asterisk-GUI Settings for ComNet Phone

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

<--- Transmitting (NAT) to 192.168.88.249:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.88.249:5060;branch=z9hG4bK-d8754z-b7b0e2881d6f8659-1---d8754z-;received=192.168.88.249;rport=5060
From: "6001"<sip:6001@192.168.88.100;transport=UDP>;tag=8a5ff34d
To: <sip:1878200@192.168.88.100;transport=UDP>;tag=as76214c7a
Call-ID: NDk3OGUwZTlmYjFiNTExNWQwZDJjMTRhM2U5MDdiOTM.
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1878200@192.168.88.100:5060>
Content-Length: 0
<------------>
-- SIP/trunk_1-00000025 is making progress passing it to SIP/6001-00000024
Audio is at 18618
Adding codec 100003 (ulaw) to SDP
<--- SIP read from UDP:202.0.179.3:5060 --->
SIP/2.0 200 OK
From: "asterisk" <sip:85235010617@192.168.88.100:5060>;tag=as29b58819
To: <sip:1878200@202.0.179.3>;tag=fa4e3fa8
CSeq: 103 INVITE
Call-ID: 2c672c914ce39bbb6f170df52d4e9103@192.168.88.100:5060
Via: SIP/2.0/UDP 192.168.88.100:5060;branch=z9hG4bK571a7528;rport=5060
Contact: <sip:1878200@202.0.179.3:5060;user=phone>
Content-Length: 200
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 7283948 7283949 IN IP4 202.0.179.3
s=Sip Call
c=IN IP4 202.0.179.3
t=0 0
m=audio 18220 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
<------------->
--- (9 headers 9 lines) ---
Found RTP audio format 0
Adding codec 100004 (alaw) to SDP
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)

2. Create a VoIP SIP Trunk

And then re-edit again to add more itemss

3. Create a dialling prefix

Edit dialplan to add outbound SIP Trunk "NWT NetTalk"

Inbound settings

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

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35^#

發表於 2013-11-8 21:57 | 顯示全部帖子

本帖最後由角色於 2013-11-8 22:00 編輯

Asterisk-GUI Settings for HKBN 2b Trunk

1. Create VoIP Trunk

Further modify the trunk by editing.

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

3. Add the default dial plan

4. Add the incoming settings:

5. Reboot the system

6. Check the system status

1093f.gif (6.43 KB)

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36^#

發表於 2014-1-1 22:51 | 顯示全部帖子

NWT NetTalk 注册和打出问题

Registration =No or No Authentication

终于周处问题所在：

就是在sip.conf，trunk definition，要加 qualify=no

就是在

[nwt-nettalk]
username=31xxxyyy
type=friend
secret=Password
qualify=no
port=5060
insecure = port,invite
host=ngn2.nwtbb.com
fromusername=31xxxyyy
fromuser=31xxxyyy
fromdomain=home1.com
dtmfmode = rfc2833
canreinvite = no
defaultexpirey=300
context = from-nwt-nettalk

複製代碼

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37^#

發表於 2014-1-2 23:47 | 顯示全部帖子

回復 49# kinuschan

你应该在users.conf，在trunk里的settings。

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38^#

發表於 2014-1-3 00:40 | 顯示全部帖子

不知道是否与版本有关？

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39^#

發表於 2014-1-22 23:11 | 顯示全部帖子

As far as I know that there are two set of hostnames for two groups of HKBN 2b lines. If one of them does not work, we can switch to another one for testing. In general, it should solve the issue.

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