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Old March 7th, 2007, 03:04 PM
huy.banh huy.banh is offline
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Default Asterisk and clients inside LAN/NAT, can call sip url outside?

Hi all,

Maybe this topic has been discussed already. If so, please help to point me where I can get the information.

I installation: Asterisk and clients, all are on LAN, behind NAT. Calling between internal clients are fine.

However, how can I have my internal clients to call outside users, i.e. sip urls on the internet. I don't think IAX is suitable in this case, since IAX allows calls to predefined domains only. Am I correct at this?

I also think of two asterisk servers, one inside and one outside. This way, each internal user needs to setup two accounts to two servers. The issue is, users need to switch between two accounts to dial internal/external urls. This would not make them much happy

Do you have a solution for this?
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Old March 7th, 2007, 06:16 PM
rvtango rvtango is offline
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Default Re: Asterisk and clients inside LAN/NAT, can call sip url outside?

hmm ... i could dial out using URL just fine on my Polycom SIP-501. Didn't really need to do anything for it as far as I remember
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Old March 8th, 2007, 10:07 PM
huy.banh huy.banh is offline
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Default Re: Asterisk and clients inside LAN/NAT, can call sip url outside?

Hi,

It doesn't work for me. I have already set externhost to FQDN of asterisk host (I do a port forward on 5060 to my * ).

One more thing, I force clients to always use outbound proxy, which is my * . If not, I *think* clients can setup outbound calls, but media stream can't pass through the NAT (I have no chance to test this yet).

I turned on sip debug. Asterisk, on receiving an outbound invite, doesn't do srv lookup, but incorrectly treats the external domain as an internal.

On console log: "Looking for huy.banh in internal (domain ekiga.net)

I have srvlookup=yes .

Could someone please help?
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Old March 8th, 2007, 10:30 PM
huy.banh huy.banh is offline
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Default Re: Asterisk and clients inside LAN/NAT, can call sip url outside?

I confused myself. "internal" in the message is context name. The log message on console should be read:

Looking for [username] in [context-name] (domain [external.tld])

Does that mean I need to do something with extensions.conf to enable sip outbound calls?
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