Sorry, didn't read your mail thoroughly - you've already tried canreinvite=no...
Next step is to get an Ethereal log from both ends and investigate what is going on with the SIP and RTP packets. Rgds Tim -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 24 March 2004 13:31 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Phones can talk to asterisk but not each other through it In article <[EMAIL PROTECTED]>, <[EMAIL PROTECTED]> wrote: > Tony, > What is the BW connectivity at the [*] box? It's lots. Much more than the broadband connections our phones are behind. File downloads to the * box from elsewhere on the internet typically go at several hundred kbytes/sec. > You may try to set the GS phones to GSM codec to reduce BW, and see if > that improves the situation. I didn't have much success using GSM, but I'll try again. Thanks for the reply... Tony > WW > ----- Original Message Follows ----- > > I posted this a week or two ago but no replies, so trying again... > > > > Summary: Two phones in different locations, each behind NAT, can > > both talk to an Asterisk server on the net, for the demo or for > > voicemail, but can't maintain a call to each other via that > > asterisk. > > > > Original post with details: > > > > I have a problem with an installation of asterisk on my colo server. > > I have a Grandstream BT102 behind a Linux NAT firewall, and my > > colleague also has one behind his. > > > > My connection is ADSL with 512k down and 256k up. My colleague's is > > Cable with 600k down and I don't know whether it's 128k or 256k up. > > > > I have the phones set up in sip.conf with nat=yes, qualify=yes and > > canreinvite=no. Each phone can successfully connect with Asterisk > > and dial the Asterisk Demo, leave and pick up voicemail, etc. > > > > However, if one phone tries to dial the other, once the called phone > > is answered, the audio starts off very stuttery and broken, and > > after a few seconds dies completely and the call gets dropped. > > > > In the asterisk log there are many entries for that time > > saying: Recv error: Resource temporarily unavailable. > > > > I am using the zaprtc timer module on the asterisk server, but in > > any case I understood that was only required for MeetMe or MOH. > > > > The server system is a Duron XP 1800, with 512MB RAM, running Fedora > > Core 1 with updates, and a standard 2.4.22 kernel that was > > recompiled only to make the RTC a module instead of compiled in (so > > I could rmmod it and then load zaprtc instead, which works fine). > > > > Can anyone suggest what things I should check or change? > > > > Cheers > > Tony > > -- > > Tony Mountifield > > Work: [EMAIL PROTECTED] - http://www.softins.co.uk > > Play: [EMAIL PROTECTED] - http://tony.mountifield.org > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > Willy Wouters > ypOne Publishing > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
