I just discovered that the SIP channel has undergone some major improvements. I'm now able to dial any SIP URL with dial, couldn't get it to work earlier, all domains had to be defined in SIP.conf. This, in addition to the SIPDOMAIN variable, makes the SIP channel even more useful. Thank you, Mark, for your additions! Now, ENUM/E.164 will propably work even better. I'll give it a try. Now, to be the documentation-pain-in-the-*** I would like to get an explanation of the autocreatepeer SIP.conf setting and functionality? It's not in sip.conf.sample yet. /Olle
On Sat, Nov 22, 2003 at 08:51:35PM +0100, Olle E. Johansson wrote:> I just discovered that the SIP channel has undergone some major > improvements.But not, alas, in the realm of NAT. Is there any possibility of removing the broken externip implementation and importing the patch I submitted that does it properly? If there are objections to the patch, please say so and I will attempt to deal with them. Maintaining a forked chan_sip.c and importing the other changes is beginning to be a bit of a headache and making it hard to keep up with CVS... -w
Olle E. Johansson wrote:> I just discovered that the SIP channel has undergone some major > improvements. > I'm now able to dial any SIP URL with dial, couldn't get it to work > earlier, > all domains had to be defined in SIP.conf....and I'm able to call any SIP URL with Xlite, with Asterisk resolving the domain part according to DNS SRV records, contacting the right SIP proxy for the DOMAIN, setting up the call. This is brilliant, a major step forward for the SIP support in Asterisk! /O
On Sat, 2003-11-22 at 21:30, asterisk@lists.styx.org wrote:> On Sat, Nov 22, 2003 at 08:51:35PM +0100, Olle E. Johansson wrote: > But not, alas, in the realm of NAT. Is there any possibility of > removing the broken externip implementation and importing the > patch I submitted that does it properly? If there are objections > to the patch, please say so and I will attempt to deal with them. >Where can I find that patch? TIA, Patrick