voip server registrations in blended calling!!

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voip server registrations in blended calling!!

Postby paladugu_sujith » Fri Jun 13, 2008 2:06 pm

hi..


if i am trying to have two different VOIP accounts

1) for outbound calling and
2) for inbound calling

i have to registers both the sip details in sip.conf rite

and configure the DID number only in extensions.conf.

is tht all i need to do in configuration part??

thnx

CHEERS
paladugu_sujith
 
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Postby mflorell » Fri Jun 13, 2008 3:33 pm

if you want to dial out through the SIP provider you will have to define it in the extensions.conf as well.
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thnx for the reply

Postby paladugu_sujith » Sun Jun 15, 2008 8:55 am

hi

i will try to be more clear.i have an vicidial setup currently running outbound calling.

i want to add inbound campiagn to it.

i knw how to configure in admin section as i have the managers manual with me..but i am confused about the dialplan(extensions.conf) and sip.conf

note: i have 2 different lines for outbound and inbound.

outbound is done by a sip voip minutes provider and
for inbound the provider gave a virtual number.

can anyone help hw do i configure tht on the same server running outbound also...??

thnx
cheers
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Postby mflorell » Sun Jun 15, 2008 1:19 pm

Do you know how the DID digits are sent to you?
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Postby paladugu_sujith » Sun Jun 15, 2008 1:36 pm

i am sorry.. i coudnt understand ur question? hw digits r sent???
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Postby mflorell » Sun Jun 15, 2008 2:13 pm

When you are called, how do the digits appear in the Asterisk CLI?
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Postby paladugu_sujith » Sun Jun 15, 2008 2:26 pm

thnx fr the reply..

ther is no cli output at all wen i call that DID no.


for outbound

username :xyz
pass: abc
domain:123.123.123.123

and for inbound my provider gave these details

username:wxy
pass: def
domain:voice.wpin.net

wen i enter sip show registry it says authentication failed.,. and wen i asked my provider about it he said to configure it correctly at ur end.

kindly help me out with the sip.conf and extensions.conf.

i tried several times frm the scratch bt in vain.

plzz provide me with a sample sip.conf and extensions.conf

thnx a lot
cheers
paladugu_sujith
 
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Postby Op3r » Wed Jun 18, 2008 4:20 am

get the sample extensions.conf and sip.conf files from your provider. We dont know whats the number your provider sends to you so its better for you to ask them to provide it.

If you still dont know how to do it or understand how to do it, I think its better to get someone who knows asterisk to do it for you.
Get paid for US outbound Toll Free calls. PM me.
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Postby paladugu_sujith » Wed Jun 18, 2008 6:17 am

hello... members..

thnk u for ur time.. i upgraded astguiclient from 2.03 to 2.04..

and it was a lot simpler to tackle errors.

nt both outbound and inbound are working perfect.


thnx
paladugu_sujith
 
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xcuse me

Postby eliasferreyra » Wed Jun 18, 2008 7:44 am

r you trying to use tha dialer for inbound calls too?

or you need only receive a incoming call to any routed extension?
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Postby paladugu_sujith » Wed Jun 18, 2008 8:52 am

hi

my inbound SIP DID provider gave me a 11 digit virtual number

164636XXXXX a username ,passwrd and domain.

the call is coming to asterisk and successfully routed to agent.

thnx to vicidial forum

the posts were really helpful.

thnx
paladugu_sujith
 
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Joined: Wed Jan 02, 2008 1:42 pm


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