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Author Topic: Question about Asterisk tickets  (Read 3512 times)
darkwizard864
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« Reply #45 on: December 22, 2012, 08:54:02 pm »

still no go
now incoming calls go directly to voicemail and caller id is unknown

here is log
pastebin.com/5pyx4vkt
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cfernandes
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« Reply #46 on: December 22, 2012, 09:31:12 pm »

well, I will reverse the order of testing.


try again please
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darkwizard864
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« Reply #47 on: December 22, 2012, 09:47:08 pm »

okay now caller id works......
but now incoming calls go directly to voice mail ext 100

here log
pastebin.com/DtYu7g07
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cfernandes
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« Reply #48 on: December 22, 2012, 09:52:55 pm »

OK .

from your logs i seed that call extensions  200  201 and 202

please check  call routes for your line
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darkwizard864
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« Reply #49 on: December 22, 2012, 09:54:38 pm »

yes i did 3 times and still direct voicemail.
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cfernandes
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« Reply #50 on: December 22, 2012, 09:59:50 pm »

you can use phones  to dial out ?   i see some congestion on  logs .

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darkwizard864
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« Reply #51 on: December 22, 2012, 10:05:26 pm »

yes well phones now work but oncreen still shows no callerid but that not a big deal
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cfernandes
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« Reply #52 on: December 22, 2012, 10:08:26 pm »

well
is working now ? on screen only not show caller id ?
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darkwizard864
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« Reply #53 on: December 22, 2012, 10:08:59 pm »

correct


2nd question how do i get it to generate voicemail users?
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cfernandes
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« Reply #54 on: December 22, 2012, 10:13:31 pm »

nice,

what are you talking about? generate the IVR with the users name?
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darkwizard864
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« Reply #55 on: December 22, 2012, 11:54:37 pm »

yes
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cfernandes
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« Reply #56 on: December 23, 2012, 01:26:08 pm »

The script is responsible for creating the db_create_dialplan.sh
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darkwizard864
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« Reply #57 on: December 23, 2012, 04:06:04 pm »

not generating the correct voice mail menu and voice mail not showing up on orbiter or webadmin.


also it shows in cdr alot of this
2012-12-23 09:32:14  SIP 10 Answer s  ANSWERED 00:01
« Last Edit: December 23, 2012, 04:07:47 pm by darkwizard864 » Logged
l3mce
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« Reply #58 on: December 23, 2012, 09:34:34 pm »

I sent you a PM cfernandes.

Is this something I need to fold into the existing source? I was asked to close the ticket, and a works for one guy is not really how that should work... or was his situation a one off problem?
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I never quit... I just ping out.
cfernandes
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« Reply #59 on: December 23, 2012, 11:20:55 pm »

L3mce

this a really a problem so that the caller ID is displayed.

for me, the caller ID is passed by p-asserted-id (Grandstream HT-503)
SPA3102 and some voip provieders  uses to sip reader (FROM) 

I must make this script better.
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