Tony Mountifield
2004-Mar-13 16:08 UTC
[Asterisk-Users] SIP Recv error when talking via asterisk
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: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org