| Author |
Message |
Glenn Robinson
Guest
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 Posted:
Fri Sep 16, 2005 3:04 am Post subject:
Asterisk and SIP |
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Hello,
I've set up asterisk and I can use it internally without any problems.
I've now got my self a phone number from Gradwell (uk) and have set up a
trunk in AMP and set incoming calls to my number to be routed through to my
softphone.
When I dial my new number from a PSTN phone I can see it logged in asterisk
with Disposition of "NO ANSWER". This is true as I get the standard "Your
call cannot be connected" message on the PSTN phone.
Any ideas why the calls are getting in to my asterisk server but then not
being routed through to my extension?
Thanks
--
Glenn |
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Ramon F Herrera
Guest
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| Posted:
Fri Sep 16, 2005 4:21 am Post subject:
Re: Asterisk and SIP |
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Glenn:
In my experience, you are not going to get many answers to such
specific question in this NG.
Your best bets are: the Asterisk Group in Google, or this forum:
http://forums.digium.com/
Then, there is the mailing list(s).
Good luck,
-Ramon
ps: the first thing you are going to be asked is to post your sip.conf
and extensions.conf files. |
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Ivor Jones
Guest
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| Posted:
Mon Sep 19, 2005 4:13 am Post subject:
Re: Asterisk and SIP |
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"Ramon F Herrera" <ramon@conexus.net> wrote in message
news:1126826507.088213.130500@f14g2000cwb.googlegroups.com
| Quote: | Glenn:
In my experience, you are not going to get many answers
to such specific question in this NG.
Your best bets are: the Asterisk Group in Google, or this
forum:
http://forums.digium.com/
Then, there is the mailing list(s).
Good luck,
-Ramon
ps: the first thing you are going to be asked is to post
your sip.conf and extensions.conf files.
|
You might want to try uk.telecom.voip as well, quite a few people there
know a lot about Asterisk (I'm not one of them..!). Unfortunately the
group is suffering from an attack of the trolls at the moment, but if you
can ignore them you might find some answers.
Ivor |
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