Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: jameswilson on April 20, 2010, 05:34:30 PM
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We currently use 3cx, but my love of sme made me investigate this contrib.
All seems to be going well but i cant get a sipgate trunk to work.
i know its me. i have followed this guide
http://www.freepbx.org/support/documentation/howtos/howto-setting-up-voip-provider-trunks/sipgate-u-k
but still no joy.
i probably shouldnt of done this but i have no also updated all modules etc from within freepbx.
Any suggestions please?
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I have sipgate trunk working with the following config
Registration String: <sipgate_number>:<sipgate_password>@sipgate.co.uk/<sipgate_number>
type=peer
host=sipgate.co.uk
insecure=very
qualify=3000
canreinvite=no
username=<sipgate_number>
fromuser=<sipgate_number>
fromdomain=
secret=<sipgate_password>
disallow=all
allow=alaw
allow=ulaw
Obviously replace <sipgate_number> and <sipgate_password> with your respective values.
This is with asterisk14 and sail-2.4.1-12, but it has been working for many previous versions.
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Hi,
I have setup Sipgate with SME server without a problem. The problem might be with your registration string. You need to add full phone number at the end <sipgate_number> field. so lets say you full sigate number is 0208 999111 where last six digits represent your sipgate username then your registration string should look something like this
Registration String: 999111:<sipgate_password>@sipgate.co.uk/0208999111
I hope that helps. :)
Muhammad
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Mine registers with out including the STD code at the end of the Reg string. I just use the main last six digits for <sipgate_number> in all locations and have no problem. I do have an 0845 number however, not that it should make any difference. Its obviously worth trying with and without STD if you have problems
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Timn
Thats sorted it.
Many thanks