Thanks Rich - Maybe I'll try the dev mailing list.
I'm not that familiar with the protocol level as well. I'm thinking its related to one of those two items (user=phone or the Contact: being blank). I've looked through all the configs and I don't seem to see any way to have the Contact fall back to parsing the SIP To: field. For now I have a macro which does this "ok" but I can't use and of the web interface tools to manage these extensions as I basically have to bypass the extension pass the "extension" in to a 2nd macro for dialing, voicemail, ec. RJ On 8/13/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
> > I'm using a wholesale voip origination provider - they don't deal with > end users. As such they have statically defined my Asterisk box on > their end - there's no registration or authentication by my system > with theirs - other than them hardcoding the destination IP of my > server in their system. > > As such I don't have a username to "register" with them. Additionally > they will be originating 100s of DIDs for me at the end of the day, so > it would be horrible to have to register all these in some way. > > Also, I don't have any "register" statements with provider A, and > somehow that seems to work out just fine as well. > > It seems to me like Asterisk doesn't like the format of what's coming > from their system. Perhaps the addition of: ";user=phone" is confusing > Asterisk? > > That's the only obvious difference, that and the Contact: header being > set to "s". > > Any other ideas? Nope, other then if I were trying to troubleshoot the issue, I'd get an ethereal trace of both A and B along with a matching sip debug. By comparing the two trace methods one should be able to rule out at least some issues and narrow the possible root cause. I don't consider myself a sip protocol expert, but I'd have to guess that Olle, Kevin, and few others on the list would be able to help interpret that output if its condensed to some reasonable size, and not summarized by selected copy/paste of portions of the traces that might miss important details. Given the level of detail and amount of analysis time needed to fully understand the issue, posting all of that to the list probably isn't going to get the wanted result. Maybe a private email to one of the more knowledgeable sip folks requesting their assistance might be helpful. If the problem turns out to be something like the ";user=phone" mentioned, I'd have to wonder if the wholesale provider would actually do anything about it though. R. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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