[Asterisk-Dev] Changes to SIP

[James Dennis]

Yes registering works fine. But as far as I understand it that is a separate
issue to an outgoing INVITE. Outgoing calls are not linked to the REGISTER.

-----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] On Behalf Of alex at pilosoft.com
Sent: 05 April 2003 10:51 PM
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] Changes to SIP


> I'm not sure how many people are using the SIP stuff but I have 
> noticed that a few changes need to be made for asterisk to correctly 
> work with a service provider running Cisco proxy. I am not very 
> familiar with SIP and basically would like comments on these points 
> before attempting to make modifications. I do not wish to break it for 
> anything else:
already possible:
register = 18007107031:xxxx at sipauth.deltathree.com/777718007107031

Rest of your mail are valid concerns. SIP stack needs a lot of work. ;)

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