I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's simply acting as a modem with one of it's LAN ports connected to
FE4 WAN on my Cisco 881.
The Cisco 881 get's a DHCP provided IP from my ISP. My LAN is part of default Vlan 1 (192.168.1.0/24). General internet connectivity is working great, I've managed to setup static NAT rules for my HTTP/HTTPS/SMTP/etc. services which are running on my LAN. I don't know whether it's worth mentioning that I've opted to use NVI NAT (
ip nat enable as opposed to the traditional
ip nat outside/
ip nat inside) setup. My reason for this is that NVI allows NAT loopback from my LAN to the WAN IP and back in to the necessary server on the LAN.
I run an Asterisk 1.8 PBX on my LAN, which connects to a SIP provider on the internet. Both inbound and outbound calls through the old setup (WAG320N providing routing/NAT) worked fine. However, since moving to the Cisco 881, inbound calls drop after around 10 seconds, whereas outbound calls work fine.
The following message is logged on my Asterisk PBX:
[Dec 9 15:27:45] WARNING: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission email@example.com for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6528ms with no response [Dec 9 15:27:45] WARNING: chan_sip.c:3670 retrans_pkt: Hanging up call firstname.lastname@example.org - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
(I know that this is quite a common issue - I've spend the best part of 2 days solid on this, trawling Google.)
I've done as I am told and checked https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions. Referring to the section "Other SIP requests" in the page linked above, I believe that the hangup to be caused by the
ACK from my SIP provider not being passed back through NAT to Asterisk on my PBX.
I tried to ascertain this by dumping the packets on my WAN interface on the 881. I managed to obtain a PCAP dump of packets in/out of my WAN interface. Here's an example of an ACK being reveived by the router from my provider:
689 21.219999 193.x.x.x 188.x.x.x SIP 502 Request: ACK sip:email@example.com |
However a SIP trace on the Asterisk server show's that there are no
ACK's received in response to the
200 OK from my PBX:
In the past, I have been strongly advised to disable any sort of SIP ALGs on routers and/or firewalls and the many posts regarding this issue on the internet seem to support this. However, I believe on Cisco IOS, the config command to disable SIP ALG is
no ip nat service sip udp port 5060 however, this doesn't appear to help the situation. To confirm that config setting is set:
Router1#show running-config | include sip no ip nat service sip udp port 5060
Another interesting twist: for a short period of time, I tried another provider. Luckily, my trial account with them is still available, so I reverted my Asterisk config back to the revision before I integrated with my current provider. I then dialled in to the DDI associated with the trial trunk and the call didn't get hung up and I didn't get the error above! To me, this points at the provider, however I know, like all providers do, will say "There's no issues with our SIP proxies - it's your firewall." I'm tempted to agree with this, as this issue was not apparent with the old WAG320N router when it was doing the NAT'ing.
I'm sure you'll want to see my running-config too:
! ! Last configuration change at 15:55:07 UTC Sun Dec 9 2012 by xxx version 15.2 no service pad service tcp-keepalives-in service tcp-keepalives-out service timestamps debug datetime msec localtime show-timezone service timestamps log datetime msec localtime show-timezone no service password-encryption service sequence-numbers ! hostname Router1 ! boot-start-marker boot-end-marker ! ! security authentication failure rate 10 log security passwords min-length 6 logging buffered 4096 logging console critical enable secret 4 xxx ! aaa new-model ! ! aaa authentication login local_auth local ! ! ! ! ! aaa session-id common ! memory-size iomem 10 ! crypto pki trustpoint TP-self-signed-xxx enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-xxx revocation-check none rsakeypair TP-self-signed-xxx ! ! crypto pki certificate chain TP-self-signed-xxx certificate self-signed 01 quit no ip source-route no ip gratuitous-arps ip auth-proxy max-login-attempts 5 ip admission max-login-attempts 5 ! ! ! ! ! no ip bootp server ip domain name dmz.merlin.local ip domain list dmz.merlin.local ip domain list merlin.local ip name-server x.x.x.x ip inspect audit-trail ip inspect udp idle-time 1800 ip inspect dns-timeout 7 ip inspect tcp idle-time 14400 ip inspect name autosec_inspect ftp timeout 3600 ip inspect name autosec_inspect http timeout 3600 ip inspect name autosec_inspect rcmd timeout 3600 ip inspect name autosec_inspect realaudio timeout 3600 ip inspect name autosec_inspect smtp timeout 3600 ip inspect name autosec_inspect tftp timeout 30 ip inspect name autosec_inspect udp timeout 15 ip inspect name autosec_inspect tcp timeout 3600 ip cef login block-for 3 attempts 3 within 3 no ipv6 cef ! ! multilink bundle-name authenticated license udi pid CISCO881-SEC-K9 sn ! ! username xxx privilege 15 secret 4 xxx username xxx secret 4 xxx ! ! ! ! ! ip ssh time-out 60 ! ! ! ! ! ! ! ! ! interface FastEthernet0 no ip address ! interface FastEthernet1 no ip address ! interface FastEthernet2 no ip address ! interface FastEthernet3 switchport access vlan 2 no ip address ! interface FastEthernet4 ip address dhcp no ip redirects no ip unreachables no ip proxy-arp ip nat enable duplex auto speed auto ! interface Vlan1 ip address 192.168.1.1 255.255.255.0 no ip redirects no ip unreachables no ip proxy-arp ip nat enable ! interface Vlan2 ip address 192.168.0.2 255.255.255.0 ! ip forward-protocol nd ip http server ip http access-class 1 ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000 ! ! no ip nat service sip udp port 5060 ip nat source list 1 interface FastEthernet4 overload ip nat source static tcp x.x.x.x 80 interface FastEthernet4 80 ip nat source static tcp x.x.x.x 443 interface FastEthernet4 443 ip nat source static tcp x.x.x.x 25 interface FastEthernet4 25 ip nat source static tcp x.x.x.x 587 interface FastEthernet4 587 ip nat source static tcp x.x.x.x 143 interface FastEthernet4 143 ip nat source static tcp x.x.x.x 993 interface FastEthernet4 993 ip nat source static tcp x.x.x.x 1723 interface FastEthernet4 1723 ! ! logging trap debugging logging facility local2 access-list 1 permit 192.168.1.0 0.0.0.255 access-list 1 permit 192.168.0.0 0.0.0.255 no cdp run ! ! ! ! control-plane ! ! banner motd Authorized Access only ! line con 0 login authentication local_auth length 0 transport output all line aux 0 exec-timeout 15 0 login authentication local_auth transport output all line vty 0 1 access-class 1 in logging synchronous login authentication local_auth length 0 transport preferred none transport input telnet transport output all line vty 2 4 access-class 1 in login authentication local_auth length 0 transport input ssh transport output all ! ! end
...and, if it's of any use, here's my Asterisk SIP config:
[general] context=default ; Default context for calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ; Specifying a port in a SIP peer definition or ; when dialing outbound calls will supress SRV ; lookups for that peer or call. directmedia=no ; Don't allow direct RTP media between extensions (doesn't work through NAT) externhost=<MY DYNDNS HOSTNAME> ; Our external hostname to resolve to IP and be used in NAT'ed packets localnet=192.168.1.0/24 ; Define our local network so we know which packets need NAT'ing qualify=yes ; Qualify peers by default dtmfmode=rfc2833 ; Set the default DTMF mode disallow=all ; Disallow all codecs by default allow=ulaw ; Allow G.711 u-law allow=alaw ; Allow G.711 a-law ; ---------------------- ; SIP Trunk Registration ; ---------------------- ; Orbtalk register => <MY SIP PROVIDER USER NAME>:firstname.lastname@example.org/<MY DDI> ; Main Orbtalk number ; ---------- ; Trunks ; ---------- [orbtalk] ; Main Orbtalk trunk type=peer insecure=invite host=sipgw3.orbtalk.co.uk nat=yes username=<MY SIP PROVIDER USER NAME> defaultuser=<MY SIP PROVIDER USER NAME> fromuser=<MY SIP PROVIDER USER NAME> secret=xxx context=inbound
I really don't know where to go with this. If anyone can help me find out why these calls are being dropped off, I'd be grateful if you could chime in! Please let me know if any further info is required.
UPDATE 10/12/2012 (1):
@GerryEgan suggested I add the following static NAT:
ip nat source static udp 192.168.1.2 5060 interface FastEthernet 4 5060
Whilst this seems to fix things up ok as far as I can tell, it would leave me open to brute force attacks on 5060.
@GerryEgan also suggested an inspect rule:
ip inspect name autosec_inspect sip
(config-if)#ip inspect autosec_inspect in on my WAN interface
Even after clearing NAT NVI translations, this alone doesn't solve the issue.
debug ip nat sip from an inbound call that gets dropped:
Note: If I try the
no ip nat service sip udp port 5060 config setting mentioned above, I don't get any debug output from
debug ip nat sip, so for now, I have the default
ip nat service sip udp port 5060 in my config - hope I'm not confusing things with this.
To make the situation worse, my static NAT NVI rules seem to be getting messed up on a reload of the router :-(. That's a separate issue though, but may have some bearing on your responses.
UPDATE 10/12/2012 (2):
When calls are hung up by Asterisk, I'm seeing the following in
Dec 10 20:02:39 <PBX> asterisk: rc_avpair_new: unknown attribute 1490026597
@GerryEgan suggested adding:
access-list 101 permit udp X.X.X.X 0.0.0.255 host 126.96.36.199 eq 5060 access-list 101 deny udp any any eq 5060 access-list 101 permit ip any any
interface FastEthernet4 ip access-group 101 in
I could see the theory so I was expecting this to work, but unfortunately not.
This is weird, as the WAG320N I have before handled SIP to the same provider, PBX, etc. with no configuration or NAT'ing required. This prompted me to rebuild the new 881, so for completeness, here's the current config:
! ! Last configuration change at 20:20:15 UTC Tue Dec 11 2012 by xxx version 15.2 no service pad service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname xxx ! boot-start-marker boot-end-marker ! ! enable secret 4 xxxx ! aaa new-model ! ! aaa authentication login local_auth local ! ! ! ! ! aaa session-id common ! memory-size iomem 10 ! crypto pki trustpoint TP-self-signed-xxx enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-xxx revocation-check none rsakeypair TP-self-signed-xxx ! ! crypto pki certificate chain TP-self-signed-xxx certificate self-signed 01 xxx quit ip gratuitous-arps ip auth-proxy max-login-attempts 5 ip admission max-login-attempts 5 ! ! ! ! ! ip domain list dmz.xxx.local ip domain list xxx.local ip domain name dmz.xxx.local ip name-server 192.168.1.x ip cef login block-for 3 attempts 3 within 3 no ipv6 cef ! ! multilink bundle-name authenticated license udi pid CISCO881-SEC-K9 sn xxx ! ! username admin privilege 15 secret 4 xxx username joe secret 4 xxx ! ! ! ! ! ip ssh time-out 60 ! ! ! ! ! ! ! ! ! interface FastEthernet0 no ip address ! interface FastEthernet1 no ip address ! interface FastEthernet2 no ip address ! interface FastEthernet3 switchport access vlan 2 no ip address ! interface FastEthernet4 ip address dhcp ip access-group 101 in ip nat enable duplex auto speed auto ! interface Vlan1 ip address 192.168.1.x 255.255.255.0 no ip redirects no ip unreachables no ip proxy-arp ip nat enable ! interface Vlan2 ip address 192.168.0.x 255.255.255.0 ! ip forward-protocol nd ip http server ip http access-class 1 ip http authentication local ip http secure-server ! ! ip nat source list 1 interface FastEthernet4 overload ip nat source list 2 interface FastEthernet4 overload ip nat source static tcp 192.168.1.x 1723 interface FastEthernet4 1723 ! ! access-list 1 permit 192.168.0.0 0.0.0.255 access-list 2 permit 192.168.1.0 0.0.0.255 access-list 101 permit udp 193.x.x.0 0.0.0.255 any eq 5060 access-list 101 deny udp any any eq 5060 access-list 101 permit ip any any ! ! ! ! control-plane ! ! banner motd Authorized Access only ! line con 0 exec-timeout 15 0 login authentication local_auth line aux 0 exec-timeout 15 0 login authentication local_auth line vty 0 4 access-class 2 in login authentication local_auth length 0 transport input all ! ! end
You'll notice that I have used
access-list 101 permit udp 193.x.x.0 0.0.0.255 any eq 5060 instead of
access-list 101 permit udp X.X.X.X 0.0.0.255 host 188.8.131.52 eq 5060. The reason being, my WAN IP is provided by DHCP, so I have no set IP to enter after
host. I hope this is a valid substitute(?).
Again, just to re-iterate - this SIP provider worked perfectly on my old 'home' Linksys WAG320N with no configuration of NAT, firewall. SIP ALG was disabled by default on that system.
Another point of interest - I'm unable to NAT PPTP inbound through this router at the moment. It appears that GRE traffic is not getting through. Whilst I know the two are not same, it could be the same underlying configuration issue that's stopping the router from NAT'ing the traffic for each of the protocols correctly - I could be barking up the wrong tree...
I've seen an improvement when commenting out
localnet=192.168.1.0/24 in Asterisk's sip.conf; I'm able to receive inbound calls without them getting hung up when this setting is omitted.
I know this could be seen as a fix to the issue, it's doesn't explain why the PBX worked with this setting for ~6 months when it's was sat behind the WAG320N as the firewall/router. Bearing this in mind, has anyone got any more ideas? Commenting out the
localnet directive doesn't sit right in my head and I'm reluctant to rely on this as a permanent solution when it worked so well with the old router.