My first post here, so a brief intro:
I'm somewhat new to Asterisk, but have been working with SIP in depth for about 3 years. I studied Asterisk for about a year and have been experimenting with it hands-on for the past month or so. I've done 6 Asterisk installs in wildly different roles/applications, some of them test systems, others in semi-production, so I know a little bit about it. I've setup voicemail, meetme, ENUM, and other Asterisk features, and I've written some AGI scripts and done some other semi-interesting tweaks.
That said, I'm curious about how others might solve the following problem. In a pure-SIP environment, if a user has an alphanumeric SIP uri, say sip:[EMAIL PROTECTED], when that user calls another SIP phone, (a real IP phone, as opposed to an ATA), via a SIP proxy, that phone can log the full URI, and 'call return' works because the SIP phone calls that URI. With Asterisk, such a call would come in with the SIP From: header (thus Caller-ID in Asterisk parlance) as something like:
From: "joe" <sip:[EMAIL PROTECTED]>;tag=as54f3792a
In this case, Asterisk doesn't know how to return the call, nor does the SIP phone, because even if the SIP phone can dial full alphanumeric URI's with some kind of a 'call return' feature, the sip:[EMAIL PROTECTED] (where 204.16.112.70 would be the IP address of the Asterisk server), isn't a valid URI and doesn't route a call to the original SIP URI: sip:[EMAIL PROTECTED]
I've thought of some tricks for handling this, and I've looked around the archives and Google searches, but haven't seen much discussion of this issue.
TIA,
David
David -
You're correct. This is an unfortunate side effect of Asterisk not really being a SIP proxy. It's a PBX replacement. Now, I understand that Olle's chan_sip2 has some of this type of feature functionality built into it, and you may want to take a look at that.
In the interim, there are some really awful, terrible, horrendous tricks you can do that might work around this problem. It involves snagging the SIP URI on the inbound call, pushing it into a database, assigning a pseudo-random number to that entry, and then keeping that mapping... forever. If a user hit the "redial" button, then the inverse would happen: your Asterisk server would dig through the database looking for the key, find the 'real' SIP URI, and re-route the call to the appropriate correct endpoint. Uuuuugly.
JT _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
