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Vicibox express install (all in one)
Version: 2.12.520a
Revision: 2418
Node Kind: directory
Schedule: normal
Last Changed Author: mattf
Last Changed Rev: 2418
Last Changed Date: 2015-11-26 18:01:37 -0600 (Thu, 26 Nov 2015)
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PfSense Firewall:
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1:1 Nat set to route WAN IP to LAN IP
TCP/UDP rule set on WAN for ports 5060-5069
UDP rule set on WAN for RTP ports 10000-20000
HTTP rule set on WAN for port 80
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Carrier Settings: (IP based registration with Vitelity)
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[vitel-outbound-nocpa]
type=friend
host=outbound.vitelity.net
context=default
trustrpid=yes
sendrpid=yes
canreinvite=no
disallow=all
allow=all
[vitel-inbound-nocpa]
type=friend
host=sip16.vitelity.net
context=trunkinbound
insecure=port,invite
canreinvite=no
allow=all
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Dial plan:
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exten => _991NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _991NXXNXXXXXX,n,Dial(SIP/${EXTEN:2}@vitel-outbound-nocpa,,ToR)
exten => _991NXXNXXXXXX,n,Hangup
exten => _881NXXNXXXXXX,1,Answer
exten => _881NXXNXXXXXX,2,Dial(${TRUNKloop}/99${EXTEN:2},,To)
exten => _881NXXNXXXXXX,3,Hangup
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Background:
New install that was configured locally in a 10.1.x.x LAN and then colocated to a 192.168.x.x LAN
Used Yast to change network config for 192.168.x.x LAN before move colocation.
Ran "ADMIN_update_server_ip.pl" to assign an appropriate IP in Vicidial for the new LAN.
Symptoms:
During initial setup in the (10.1.x.x LAN) I was able to register the trunk with Vitelity, connect my soft phone via the LAN and make an outbound call without issue.
Once the server was moved to the colocation facility I can no longer complete an outbound call with a soft phone connecting form the WAN.
I can do the following:
- Register with Vitelity (confirmed via "show sip peers")
- Register my soft phone via the WAN (confirmed via "show sip peers")
- Dial an Outbound number from my soft phone and have the phone ring
Root Issue:
When I answer the outbound call, the call is dropped.
See the "Retransmission error" in the CLI output.
CLI Output: (number is masked)
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- Code: Select all
[Dec 1 15:53:13] == Using SIP RTP CoS mark 5
[Dec 1 15:53:13] -- Executing [991402xxxxxxx@default:1] AGI("SIP/100-00000034", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 1 15:53:13] -- <SIP/100-00000034>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 1 15:53:13] -- Executing [991402xxxxxxx@default:2] Dial("SIP/100-00000034", "SIP/1402xxxxxxx@vitel-outbound-nocpa,,ToR") in new stack
[Dec 1 15:53:13] == Using SIP RTP CoS mark 5
[Dec 1 15:53:13] -- Called SIP/1402xxxxxxx@vitel-outbound-nocpa
[Dec 1 15:53:14] -- SIP/vitel-outbound-nocpa-00000035 is ringing
[Dec 1 15:53:17] -- SIP/vitel-outbound-nocpa-00000035 answered SIP/100-00000034
[Dec 1 15:53:23] WARNING[5537]: chan_sip.c:3822 retrans_pkt: Retransmission timeout reached on transmission 78101NWRjM2VmNmUyZDNhZjI1N2M2YTVmY2E4MGZjMWE0Y2E for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Dec 1 15:53:23] WARNING[5537]: chan_sip.c:3851 retrans_pkt: Hanging up call 78101NWRjM2VmNmUyZDNhZjI1N2M2YTVmY2E4MGZjMWE0Y2E - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Dec 1 15:53:23] -- Executing [h@default:1] AGI("SIP/100-00000034", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----ANSWER-----10-----6") in new stack
[Dec 1 15:53:23] -- <SIP/100-00000034>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----ANSWER-----10-----6 completed, returning 0
[Dec 1 15:53:23] == Spawn extension (default, 991402xxxxxxx, 2) exited non-zero on 'SIP/100-00000034'
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Things I've Tried:
- Re-create the firewall and NAT rules and delete states tables.
- Added STUN server to soft phone config.
- Opened ALL PORTS to the server in firewall rules.
SOLUTION
Turns out when you switch IP address the sip.conf is NOT updated even if you run the ADMIN_update_server_ip.pl command.
Manually updated the /etc/asterisk/sip.conf file to the new IP address and everything worked just fine.