[asterisk-gui] Testers Needed

Andrew Latham lathama at gmail.com
Tue May 10 08:20:24 CDT 2011


On Tue, May 10, 2011 at 1:59 AM, bob yang <learnin9 at gmail.com> wrote:

> hi,Andrew Latham:
>    Thanks for your response! sorry my poor written english!
>     OK,I write my testing process......
>   * first step:*
>     (1): add the SIP trunk and try to register the remote SIP
> softswitch(another asterisk system.the remote SIP port is 5060)
>     (2): add the dial plan,  _1XX to dial the other system's sip phone.
>     (3): add a sip phone and make it to register the sip system(A)
>     (4): call 100.the two call can be connted successfully.
>     please see the attachment asterisk_5060.txt
> *   Second step:*
>     (1): Make the remote SIP system port 5080
>     (2): Modify the SIP trunk port 5080 and "Apply changes". (Yes, it can
> register the remote sip sysyem and the remote port is 5080)
>     (3): call 100, I found the SIP system send "480" error messages
>      Please see the attachment asterisk_5080.txt
>
>    Just modified the SIP register port not 5060, the call was failed......
>    In  the asterisk_5080.txt file , I found the error messages, maybe it's
> the reason! but i do not know the reason.....
>    [May 10 13:37:07] WARNING[14650]: chan_sip.c:3094 create_addr: No such
> host: trunk_1
>    [May 10 13:37:07] DEBUG[14650]: chan_sip.c:17346 sip_request_call: Cant
> create SIP call - target device not registered
>    Thank you!
>
>
>
> 2011/5/8 Erin Spiceland <espiceland at digium.com>
>
>> > I use the GUI on over 60 servers. I am more than willing to help test.
>> > I have servers with up to 3000 users and use every feature. My
>> > question is how do I help, is there a place you want feedback?
>>
>> "Works for me" messages can go to the list. If you find bugs, please enter
>> them in the issue tracker at http://issues.asterisk.org. Thanks!
>>
>> Erin
>
>
Bob, thanks for the input...

If / When you have a config issue that does not work in Asterisk, report it
as a configuration conflict.  For example [1] is a conflict of the GUI with
its own settings.  Add an issue and what you think is supposed to happen.
 For a SIP port change you can show it working in a normal Asterisk
installation first then add the GUI.

1. https://issues.asterisk.org/view.php?id=18181

-- 
~~~ Andrew "lathama" Latham lathama at gmail.com ~~~
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