[Asterisk-Users] multiple sip accounts from same sip registrar

trixter http://www.0xdecafbad.com trixter at 0xdecafbad.com
Tue May 17 11:13:38 MST 2005


Have you run into the problem where calls inbound from any of the
numbers will all goto the same context even if each as a seperate
context defined in sip.conf ?

I have that problem with a couple different providers (which makes me
think its not the providers) and may even be related to this issue, or
maybe not (I have multiple providers with the same proxy and all show in
sip show peers and sip show registry as unique entries).



On Tue, 2005-05-17 at 17:59 +0100, Matt Scott wrote:
> Dear all,
>  
> I have an asterisk sip issue which I don't believe is unique.
> I use a registrar (sipgate.co.uk) where I have 3 different accounts.
> These accounts provide me with three seperate local phone numbers
> which allow me to allocate them to seperate users.
> By using just one of these accounts I can set asterisk up to send and
> receive calls no problem.
> However, when I start to introduce an additional account I start to
> run into problems.
>  
> if I do a 'sip show peers' with a good config I think it may outline
> the problem
>  
> sip show peers
> Name/username              Host            Dyn Nat ACL Mask
> Port     Status    
> 1005/1005                  (Unspecified)    D          255.255.255.255
> 0        Unmonitored
> 1004/1004                  (Unspecified)    D          255.255.255.255
> 0        Unmonitored
> 1003/1003                  (Unspecified)    D          255.255.255.255
> 0        Unmonitored
> 1002/1002                  10.0.0.52        D          255.255.255.255
> 5060     Unmonitored
> 1001/1001                  10.0.0.51        D          255.255.255.255
> 5060     Unmonitored
> sipgate1/321****           217.10.79.219        N      255.255.255.255
> 5060     OK (52 ms)
>  
> I think it maybe a host specific ip address which must be in a table
> somewhere in asterisk.
> I have tried setting it up as a peer and dynamic but still no joy.
>  
> Is there a limitation to this within asterisk. I have provided a
> sip.conf below (adjusted), will I need to implement a SER box (more
> things to learn which is all good provided it sorts my problem)
>  
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> ;
> register => *********.******@sipgate.co.uk/*******
> register => ***:***********@sipgate.co.uk/******
> [sipgate1]
> type=friend
> context=from-sipgate1
> fromuser=******
> username=****
> authuser=*****
> secret=******
> host=sipgate.co.uk
> fromdomain=sipgate.co.uk
> nat=yes
> dtmfmode=info
> qualify=yes
> insecure=very
> canreinvite=no
> ;
> [sipgate2]
> type=friend
> context=from-sipgate2
> fromuser=*********
> username=******
> authuser=*******
> secret=*********
> host=sipgate.co.uk
> fromdomain=sipgate.co.uk
> nat=yes
> dtmfmode=info
> qualify=yes
> insecure=very
> canreinvite=no
> ;
> [1001]
> type=friend
> username=1001
> secret=*****
> host=dynamic
> dtmfmode=rfc2833
> context=from-sipphones
> ;mailbox=1001
> allow=alaw
> allow=ulaw
>  
> kindest regards
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-- 
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605   Germany +49 801 777 555 3402
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