NAT Problem... maybe?

Asterisk Asterisk Support
1

Hi,

I have an Asterisk 11 system out on the internet and 2 Digium D40 phones connecting to it on seperate internet connections and ISP’s

The first phone connects fine to Asterisk through my home internet connection and NAT
The secound phone is on a company fiberconnection (and other then NAT they say there should be no restrictions in place)

[2016-02-02 22:37:08] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16' is now Reachable. (35ms / 2000ms) [2016-02-02 22:37:11] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16-2' is now Reachable. (34ms / 2000ms) [2016-02-02 22:38:12] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16' is now UNREACHABLE! Last qualify: 35 [2016-02-02 22:38:15] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16-2' is now UNREACHABLE! Last qualify: 34 [2016-02-02 22:38:27] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16-2' is now Reachable. (36ms / 2000ms) [2016-02-02 22:38:36] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16' is now Reachable. (56ms / 2000ms) [2016-02-02 22:41:42] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16' is now UNREACHABLE! Last qualify: 38 [2016-02-02 22:42:02] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16-2' is now UNREACHABLE! Last qualify: 35 [2016-02-02 22:42:08] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16' is now Reachable. (36ms / 2000ms) [2016-02-02 22:42:12] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16-2' is now Reachable. (39ms / 2000ms) [2016-02-02 22:43:12] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16' is now UNREACHABLE! Last qualify: 36 [2016-02-02 22:43:16] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16-2' is now UNREACHABLE! Last qualify: 39 [2016-02-02 22:43:28] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16-2' is now Reachable. (35ms / 2000ms) [2016-02-02 22:43:36] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16' is now Reachable. (34ms / 2000ms) [2016-02-02 22:45:16] WARNING[1601]: chan_sip.c:4092 retrans_pkt: Timeout on 2cb77bdebc1c8f530e71fb335ef488e3 on non-critical invite transaction. [2016-02-02 22:46:42] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16' is now UNREACHABLE! Last qualify: 34 [2016-02-02 22:47:02] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16-2' is now UNREACHABLE! Last qualify: 35 [2016-02-02 22:47:08] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16' is now Reachable. (36ms / 2000ms) [2016-02-02 22:47:12] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16-2' is now Reachable. (36ms / 2000ms) [2016-02-02 22:48:12] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16' is now UNREACHABLE! Last qualify: 36 [2016-02-02 22:48:16] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer '8f16-2' is now UNREACHABLE! Last qualify: 36 [2016-02-02 22:48:28] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16-2' is now Reachable. (35ms / 2000ms) [2016-02-02 22:48:36] NOTICE[1601]: chan_sip.c:23760 handle_response_peerpoke: Peer '8f16' is now Reachable. (36ms / 2000ms)
But the other phone seams to have problems. It does register with the Asterisk server
But it keeps falling out and. If I make a call from the secound phone, the other phone rings, but no audio either direction, and Asterisk terminates the call after a few secounds.
If I call the secound phone it rings and the same problem with audio and disconnect.

Setting SIP debug on I can see Asterisk keeps retransmitting

[code]Retransmitting #1 (NAT) to 2.109.91.226:1029:
OPTIONS sip:8f16-2@2.109.91.226:1041;ob SIP/2.0
v: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK0d2c9dd2;rport
Max-Forwards: 70
f: “asterisk” sip:asterisk@172.31.1.100;tag=as097aae53
t: sip:8f16-2@2.109.91.226:1041;ob
m: sip:asterisk@172.31.1.100:5060
i: 582477336b129e11024d4c054d882f7f@172.31.1.100:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Tue, 02 Feb 2016 21:36:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
l: 0

Really destroying SIP dialog ‘9Xvo1aU26mhZbY5PVnBpA5LXRewxPRxM’ Method: REGISTER
Retransmitting #2 (NAT) to 2.109.91.226:1029:
OPTIONS sip:8f16-2@2.109.91.226:1041;ob SIP/2.0
v: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK0d2c9dd2;rport
Max-Forwards: 70
f: “asterisk” sip:asterisk@172.31.1.100;tag=as097aae53
t: sip:8f16-2@2.109.91.226:1041;ob
m: sip:asterisk@172.31.1.100:5060
i: 582477336b129e11024d4c054d882f7f@172.31.1.100:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Tue, 02 Feb 2016 21:36:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
l: 0

Retransmitting #3 (NAT) to 2.109.91.226:1029:
OPTIONS sip:8f16-2@2.109.91.226:1041;ob SIP/2.0
v: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK0d2c9dd2;rport
Max-Forwards: 70
f: “asterisk” sip:asterisk@172.31.1.100;tag=as097aae53
t: sip:8f16-2@2.109.91.226:1041;ob
m: sip:asterisk@172.31.1.100:5060
i: 582477336b129e11024d4c054d882f7f@172.31.1.100:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Tue, 02 Feb 2016 21:36:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
l: 0

Retransmitting #4 (NAT) to 2.109.91.226:1029:
OPTIONS sip:8f16-2@2.109.91.226:1041;ob SIP/2.0
v: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK0d2c9dd2;rport
Max-Forwards: 70
f: “asterisk” sip:asterisk@172.31.1.100;tag=as097aae53
t: sip:8f16-2@2.109.91.226:1041;ob
m: sip:asterisk@172.31.1.100:5060
i: 582477336b129e11024d4c054d882f7f@172.31.1.100:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Tue, 02 Feb 2016 21:36:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
l: 0

[2016-02-02 22:37:01] NOTICE[1601]: chan_sip.c:29767 sip_poke_noanswer: Peer ‘8f16-2’ is now UNREACHABLE! Last qualify: 37
Really destroying SIP dialog ‘582477336b129e11024d4c054d882f7f@172.31.1.100:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 2.109.91.226:1029:
OPTIONS sip:8f16@2.109.91.226:1041;ob SIP/2.0
v: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK202350aa;rport
Max-Forwards: 70
f: “asterisk” sip:asterisk@172.31.1.100;tag=as3670be33
t: sip:8f16@2.109.91.226:1041;ob
m: sip:asterisk@172.31.1.100:5060
i: 5374fae647e435e43869d34521875e1c@172.31.1.100:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Tue, 02 Feb 2016 21:37:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
l: 0
[/code]

Anyone have any idea why the first phone works but the secound dosent?
And If it is a NAT issue, then how to get around it?

Sip show peers:

Name/username Host Dyn Forcerport Comedia ACL Port Status Description 8f16/8f16 2.109.91.226 D Yes Yes 1029 OK (35 ms) Digium D40 - Office 8f16-2/8f 2.109.91.226 D Yes Yes 1029 OK (36 ms) Digium D40 - Office 00b5/00b5 188.182.178.110 D Yes Yes 5060 OK (55 ms) Digium D40 - home 00b5-2/00 188.182.178.110 D Yes Yes 5060 OK (55 ms) Digium D40 - Home

Kind Regards,
Jonas

2

Unfortunately you appear to have done everything right from the Asterisk side, and it really is the device doing NAT either dropping the NAT mapping EXTREMELY soon so you can’t keep contacting the device… or in the case of RTP not doing symmetric RTP (instead of establishing a mapping back it just drops the RTP traffic).

3

Yes that was my assumption to.

Just wanted to check if anyone else out there had seen a simular issue before.
I had one of the companies network guys check it today and he thinks it’s a Firewall issue with there ISP(who controls the firewall). So I have to contact them tomorrow to have a look at it.

4

Typically firewalls come with a feature called SIPALG ask them if this is enabled on the firewall/router. If it is ask them to disable it