Hello, I am running a new Asterisk 13.6.0 installation on a virtual machine and am experiencing random dropped calls. While talking, the line simply goes dead, and the user’s phone returns to the idle screen. Note that many times this does NOT happen, even for calls of long duration. And it can drop the call within the first 60 seconds or several minutes into the conversation.
Also, when this happens, and at other random times, Asterisk reports that all extensions (both internal and external) are “Unreachable”. “sip show peers” also shows all extensions unreachable.
After a few seconds, everything resumes and Asterisk reports that the extensions are now “Reachable” again.
I am looking for advice on how to troubleshoot this issue. Any suggestions greatly appreciated! Thank you!
Here are the configuration details:
Network Map: gliffy.com/go/publish/9273973
TRUNK:
Twilio.com Elastic SIP Trunk
ASTERISK SERVER:
Asterisk 13.6.0
Virtual Host: XenServer 6.2
Virtual Guest: Centos 7 64-bit, 1GB RAM, 32GB HD, 1Gbit ethernet NIC
Static LAN IP address assigned: 192.168.1.254
Using iptables firewall - ports opened for SIP, SIP-RTP, etc.
INTERNET CONNECTION:
Time Warner, 50Mbit x 5Mbit cable internet
Dynamic public IP (but the IP never changes - has been the same for 2+ months)
Internet facing Router: Cisco RVS4000 with latest firmware and the appropriate ports forwarded to the Asterisk server.
Stability: 38ms avg, 0% packet loss on over 10,000+ packets (pinging 8.8.8.8 google DNS server)
Speed tests: Over 60Mbit downloads, 6+ Mbit uploads
PHONES:
Grandstream GXP-2000
Assigned static LAN IP addresses (192.168.1.200 and above)
Core verbosity set to 5. output during a dropped call follows. Phone numbers replaced with *.
pbx1*CLI> -- SIP/twilio-0000002f answered SIP/100-0000002e
-- SIP/twilio-0000002f answered SIP/100-0000002e
pbx1*CLI> -- Channel SIP/twilio-0000002f joined 'simple_bridge' basic-bridge <0f5aa636-a233-4e1b-b5cf-91661544bc4a>
-- Channel SIP/twilio-0000002f joined 'simple_bridge' basic-bridge <0f5aa636-a233-4e1b-b5cf-91661544bc4a>
pbx1*CLI> -- Channel SIP/100-0000002e joined 'simple_bridge' basic-bridge <0f5aa636-a233-4e1b-b5cf-91661544bc4a>
-- Channel SIP/100-0000002e joined 'simple_bridge' basic-bridge <0f5aa636-a233-4e1b-b5cf-91661544bc4a>
pbx1*CLI> [Oct 26 14:05:02] WARNING[19263]: chan_sip.c:4009 retrans_pkt: Retransmission timeout reached on transmission 3103503ad5cbe2cb@192.168.1.200 for seqno 25125 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 10401ms with no response
[Oct 26 14:05:02] WARNING[19263]: chan_sip.c:4038 retrans_pkt: Hanging up call 3103503ad5cbe2cb@192.168.1.200 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Oct 26 14:05:02] WARNING[19263]: chan_sip.c:4009 retrans_pkt: Retransmission timeout reached on transmission 3103503ad5cbe2cb@192.168.1.200 for seqno 25125 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 10401ms with no response
[Oct 26 14:05:02] WARNING[19263]: chan_sip.c:4038 retrans_pkt: Hanging up call 3103503ad5cbe2cb@192.168.1.200 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
pbx1*CLI> -- Channel SIP/100-0000002e left 'simple_bridge' basic-bridge <0f5aa636-a233-4e1b-b5cf-91661544bc4a>
== Spawn extension (internal, *******, 4) exited non-zero on 'SIP/100-0000002e'
-- Channel SIP/100-0000002e left 'simple_bridge' basic-bridge <0f5aa636-a233-4e1b-b5cf-91661544bc4a>
== Spawn extension (internal, *******, 4) exited non-zero on 'SIP/100-0000002e'
pbx1*CLI> -- Channel SIP/twilio-0000002f left 'simple_bridge' basic-bridge <0f5aa636-a233-4e1b-b5cf-91661544bc4a>
-- Channel SIP/twilio-0000002f left 'simple_bridge' basic-bridge <0f5aa636-a233-4e1b-b5cf-91661544bc4a>