[Asterisk-Users] SIP Recv error when talking via asterisk
tony at softins.clara.co.uk
Sat Mar 13 16:08:59 MST 2004
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?
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
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