I’ve had asterisk 1.0.x running for about two years now. Very simple setup that registers to my SIP provider, and distributes calls to a few extensions and voicemail. One extension is a Sipura SPA-2000, and the other is X-lite. Both are behind NAT, and my asterisk server is on a public IP.
Everything has been working great for the last two years – no problems whatsoever.
Recently, I tried upgrading to 1.2.10. I cleared out /usr/lib/asterisk/modules and /etc/asterisk, installed 1.2.10 from source, and then copied over my existing sip.conf, extensions.conf and voicemail.conf. At first I thought the upgrade had gone very smoothly – the extensions still registered fine, asterisk registers to my SIP provider fine, and outbound calls from the extensions also working fine. However, when an inbound call comes in from my SIP provider and asterisk tries to ring the extensions, it fails somehow. Call goes straight to voicemail. Here’s what I see in the log:
-- Executing Dial("SIP/1095-09f8e878","SIP/meehan-home&SIP|30") in new stack -- Called meehan-home
Jul 28 13:32:19 WARNING: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission firstname.lastname@example.org for seqno 102 (Critical Request)
Jul 28 13:32:19 WARNING: chan_sip.c:1234 retrans_pkt: Hanging up call email@example.com - no reply to our critical packet.
== Everyone is busy/congested at this time (2:0/0/2)
If I turn on SIP debug, I see the INVITE and retries going out to the extension, but the extension never replies. Here’s the relevant section from my sip.conf:
If I revert back to Asterisk 1.0.11, everything works fine again. Any ideas on what might be going on here?