I think you chose the right solution in your case. It might doesn't "feel" right if you corrected source code, I think.
23.08.2015, 20:53, "Sam" <[email protected]>: > On 08/21/2015 12:52 AM, Sam wrote: >> Hello, >> >> I have what I would think would be a common situation: I run asterisk at >> home simply as a land line. I started a new job working remotely and >> they gave me a SIP account with user name, domain, and proxy. I've never >> had to deal with sip domains before. My user '[email protected]' is >> handled by a 3rd party provider: 'sip.provider.com' and my local domain >> on my asterisk box is the hostname 'mypbxdomain.com'. >> >> My normal extension I use for everything is just '111'. I figured the >> best way of joining my asterisk box was to just hard code in the >> extensions I would need to dial for my remote office work (there are >> only a couple of extensions so shouldn't be a big deal). >> >> However I struggled to get authentication working for outgoing calls to >> the few new extensions at the remote office through their provider. >> Looking at debug logs it was clear that the sip 'To' address was wrong. >> It had the provider: "To: <sip:[email protected]>" instead of the >> domain which should look like: "To: <sip:[email protected]>" (right?) >> >> In the end, after hours of googling, reading the docs on sip.conf >> several times revealed a little spoke of '!' dialplan option. Simply >> changing my dialplan from 'Dial(SIP/workphone/${EXTEN})' to >> 'Dial(SIP/workphone/${EXTEN}!${EXTEN}@4354766787.com)' fixed the issue. >> >> But this seems really hackish. Is this the right/only way? Or is just >> having a provider and mismatched domains not really the norm? >> >> I have an an anonymized log here: http://tinyurl.com/ouy2ajr >> >> Regards, >> Sam > > So since no one has responded, this either means one of two things: > Either I am an idiot and missed something obvious and therefore no one > wants to deal with me. Or this is indeed not something typical of an > asterisk config. > > Can someone at least point me to which? :) > As I mentioned, everything is working, it just doesn't "feel" right. > > Regards, > Sam > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
