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16# 跳轉到 »
發表於 2010-8-14 23:10 | 只看該作者
Since you are using Digium Asterisk-GUI to connect your 2b service, you may go to the HKEPC forum to check the thread 【ATCOM IP-01 + FXO】for the codes for 2b registration.

YH

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ckleea

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17#
發表於 2010-8-14 23:15 | 只看該作者
no, I use the registration string in sip.conf as showed by you above.

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18#
發表於 2010-8-15 15:58 | 只看該作者
How is it going?  Are you able to make outbound calls now?

YH

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ckleea

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19#
發表於 2010-8-15 16:33 | 只看該作者
Still 603 error.

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20#
發表於 2010-8-15 19:40 | 只看該作者
Are you using Digium Asterisk-GUI or using Asterisk Programming Language in configuring your 2b?

YH

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ckleea

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21#
發表於 2010-8-15 21:01 | 只看該作者
APL now. I do not configure HK2b as VOIP trunk under GUI

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22#
發表於 2010-8-15 21:24 | 只看該作者
You meant you use using APL in a machine with Asterisk-GUI 2.0 installed right?

YH

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ckleea

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23#
發表於 2010-8-15 22:00 | 只看該作者
Yes. To generate the code, I would use APL which is faster and can test easily. It is slow to test with GUI.

I use WINSCP and putty to edit and login asterisk console.
When finished editing, a reload command at console

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ckleea

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24#
發表於 2010-8-16 12:45 | 只看該作者
Should I go back to Asterisk 1.4?

I have try to put the registration into IP01 but not working. The next will be into a DD-WRT router which is limiting in function.

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25#
發表於 2010-8-16 14:02 | 只看該作者
For IP-01， lttliang has one and he put his 2b in ip-01 without any problems in both outbound and inbound calls.

YH

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ckleea

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26#
發表於 2010-8-16 15:10 | 只看該作者
I disable the 2b registration in asterisk and use their software, I think it may be due to the of s21.hkbntel.net and s22.hkbntel.net

When I fix to s21.hkbntel.net, it keeps return to s22.hkbntel.net. Somethings wrong not sure what to do next.

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ckleea

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27#
發表於 2010-8-16 17:19 | 只看該作者
HKBN also provides a VPN client to use their soft phone. I am actively looking up the information

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28#
發表於 2010-8-16 18:31 | 只看該作者
If you still have problem, I may help you by letting me know the credentials of your 2b again. I shall implement in my Asterisk 1.4 server in Hong Kong. If it works for both inbound and outbound calls in Asterisk 1.4, then I believe there may be some settings in Asterisk 1.6 that you have figured out what they are.

YH

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ckleea

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29#
發表於 2010-8-16 19:03 | 只看該作者
Will try this one tonight to look into the sip message

http://sipx-wiki.calivia.com/ind ... Messages_to_display

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30#
發表於 2010-8-16 19:37 | 只看該作者
It seems that you are trying your very best to fix the 2b problem yourself. I wish you could make your 2b system work.

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