I would have thought so; however, interestingly my VoiceGlobe account (in the form of @<e-mail_account.com>) registers just fine to its server with no quotes from my asterisk-1.6.x. Honestly, I have no idea why mine works; however, if no quotes don’t work for OP, it certainly won’t hurt to try with quotes to see if that will resolve the issue.
See my post above that the @ works just fine on my asterisk-1.6.x with my VoiceGlobe account.[/quote]
Well did not work with mine. The thing that saved me is the fact the last part of the register string can be a context. In that context you can define all kinds of things like fromdomain. It can give you the same registration SIP packet as in 1.4.
Still the question remains why so significant changes are introduced in between 1.4 and 1.6 ?
Did you compare the codes from 1.4 to 1.6? Unless you have done such a comparison, there isn’t any changes between 1.4 to 1.6 w.r.t the @ that I am aware of. Afterall, my asterisk-1.6.x, whose sip.conf file with my VoiceGlobe account was inherited from my asterisk-1.4.x with one change to replace the depricated username with the new defaultuser, still works fine with the @ in its username. Had there been any changes, perhaps such changes would either have caused the problems (which isn’t the case) or didn’t cause any problems (which is the case) to my existing sip.conf file. May be, it is best for you to post your sip.conf file here for readers to digest and come up with some sort of solution.
It works because your sip provider does not care about the REGISTER line.
May be, it is best for you to post your sip.conf file here for readers to digest and come up with some sort of solution.[/quote]
You did not read my post where I explained that I got it already working after using context as the last part of the register string.
It works because your sip provider does not care about the REGISTER line.[/quote]
I don’t think so. I believe VoiceGlobe supports @ username. Otherwise, I will have no need to use the @. One thing you may want to try is to create a free account on VoiceGlobe, and see if you still encounter the same problem. Also, I wouldn’t mind if the VoSP in question has offer a free SIP P2P account so I can try it.
I did. Perhaps, you didn’t read my post saying that my VoiceGlobe configuration was copied directly from asterisk-1.4.x to asterisk-1.6.x with no modification w.r.t @ username and it still works!