search for: channel_sip

Anyone that knows what support we have for SUBSCRIBE? There's some code in chan_sip.c that do something. And there's a comment on SNOM 200. Curious. /Olle

...ptions like nat= canreinvite= reinvite= and qualify= you > should be able to get most NATed phones to work.. As for Asterisk as a > Client I havent really had a lot of experience with that so I am not > really sure what is needed.. > > > I think a lot of things can be added to channel_sip.c without doing > the > > channel_sip.c-ng++ rearchitecture. This said standing on thin ice > since > > I haven't got full insight into how channel_sip works, source-code > wise. > > I can't really help there either.. I know squat about C coding.. and the >...

I'm trying to use early-dial. Here, all hardware PBX have it. You dial numbers and, as soon as you have a matching dialplan entry, you get throught. I my Grandstream I enabled early-dial. And when I put exten=91,1,Milliwatt in my dialplan then it works as expected. Also, when I call other SIP or IAX phones it works. Hurray! But how can I get it working with external lines? In my

I've tried to find documentation on if Asterisk supports DNS SRV records for sip servers. Reading the source of channel_sip.c it seems not: hp = gethostbyname(hostname); if (!hp) { ast_log(LOG_WARNING, "Host '%s' not found at line %d\n", hostname, lineno); return -1; } This is in the register part. Am I correct in this or have I missed something? /Olle

...ving the issue described in this question: http://lists.digium.com/pipermail/asterisk-users/2005-May/099075.html Does anybody has an insight? I guess Asterisk is trying to match the combination IP:Port, but in H223 this changes call by call. There is no way to add "insecure=port" like in channel_sip.