Hi Michael, thanks for you answer. Comments below.
Am Donnerstag, den 13.01.2005, 12:38 +0100 schrieb michael koehler: > inline > > On Jan 10, 2005, at 10:12 PM, Christian Peter wrote: > > > > - If I call outside (with Nikotel to German Telekom) there is a remote > > hangup after 2 minutes. I've seen other people posting this but nothing > > helped. I luckily managed to get around this issue with the following > > workaround: The provider section should only contain disallow=all and > > then only allow=gsm. If I add allow=alaw ..... > > After 60 seconds nikotel send a reINVITE to your box. If your box does > not respond > then the call gets cleared after 120 seconds. I do not know why this is > up to the codec > order of * > > I've got more codec problems. See below. > > - I sniffed the traffic and came to another strange issue. From time to > > time asterisk sends a OPTIONS packet (even before REGISTER). This > > Seems that * keeps routers WAN port this way > > > packets have a From header which looks like this: > > <sip:[EMAIL PROTECTED]> > > Nikotel does of course not recognize this address and sends a "Call leg > > or transaction does not exist". Is this a bug or intended behaviour? > > Looks like the OPTIONS request happen outside of an dialog. Ok I forgot to give externip=xyz and srvlookup=yes in the sip.conf. Now it uses the right ip address but still asterisk as username. > > > > > - No internal Nikotel call (phone number beginning with 99) reaches my > > friends (which have similar sip.conf and extensions.conf). Somewhere I > > read that the section must be named like the host > > "calamar0.nikotel.com" > > so that asterisk finds it. It didn't help. Did someone manage to get > > this working? > > There is(should be) a 302 Response fix in the current CVS > I tried it with current 1.0 CVS (and 1.0.2 and 1.0.3 :)). The redirect now works if one specifies the ip of calamar0.nikotel.com as section name. Now there are two possibilities after the phone accepts the call: - If I remove the disallow=all, allow=xyz stuff from sip.conf asterisk gives no error message but i can't hear something (Nikotel testnumber: 999 900 900 900) - If I put disallow=all, allow=gsm in the sip.conf asterisk tells me that the codecs are incompatible. I'm now giving it up. I spent almost a week on the 99er issue and still no luck. If someone has Nikotel FULL working I would be glad to see his/her configs. Greetings Christian > > Michael > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users