[VoIP] Help...............

[Ian Jolly]

Hi All

Help ! - we are still trying to set up an ATA working off my * tandem (as a 
'remote' extension) to an FXS port on a Draytek 2600V router. We are using 
SIP. The other SIP peer 'SPA3102' is working OK on my internal network.

We've got as far as me being able to ring the phone at the ATA  identified 
as 'jgriffiths'.
My fixed IP is 81.174.170.48
'jgriffiths' fixed IP is 80.176.242

However I get the following - I'm not shouting - it is just easier to see my 
comments and the cut and paste from the CLI> prompt.

AT THE CLI> prompt - I GET A MESSAGE LIKE THIS - ANY CALL TO THE ATA IS CUT 
OFF -

Oct 15 23:35:18 NOTICE[2490]: chan_sip.c:11328 sip_poke_noanswer: Peer 
'jgriffiths' is now UNREACHABLE! Last qualify: 134
Destroying call '791c82153ca0c38868892e5438d88da2 at 81.174.170.48'

A MESSAGE LIKE THIS IS SENT EVERY FEW SECONDS WITH A DIFFERENT DIGIT BEFORE 
THE # SIGN-

Retransmitting #4 (NAT) to 80.176.242.9:5060:
OPTIONS sip:jgriffiths at 80.176.242.9 SIP/2.0
Via: SIP/2.0/UDP 81.174.170.48:5060;branch=z9hG4bK3d3f9f80;rport
From: "asterisk" <sip:asterisk at 81.174.170.48>;tag=as162e4695
To: <sip:jgriffiths at 80.176.242.9>
Contact: <sip:asterisk at 81.174.170.48>
Call-ID: 2e7c5657365eb28d67a552756a95ba33 at 81.174.170.48
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 15 Oct 2006 22:45:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
Destroying call '2e7c5657365eb28d67a552756a95ba33 at 81.174.170.48'

EVENTUALLY A MESSAGE LIKE THIS IS RECEIVED

Oct 15 23:35:56 NOTICE[2490]: chan_sip.c:9689 handle_response_peerpoke: Peer 
'jgriffiths' is now REACHABLE! (83ms / 2000ms)
Destroying call '071faaca10c99ee53e73b39c261e2472 at 81.174.170.48'

THEN MORE -

Retransmitting 14 (NAT) to 80.176.242.9:5060:
OPTIONS sip:jgriffiths at 80.176.242.9 SIP/2.0
Via: SIP/2.0/UDP 81.174.170.48:5060;branch=z9hG4bK3d3f9f80;rport
From: "asterisk" <sip:asterisk at 81.174.170.48>;tag=as162e4695
To: <sip:jgriffiths at 80.176.242.9>
Contact: <sip:asterisk at 81.174.170.48>
Call-ID: 2e7c5657365eb28d67a552756a95ba33 at 81.174.170.48
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 15 Oct 2006 22:45:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
Destroying call '2e7c5657365eb28d67a552756a95ba33 at 81.174.170.48'

UNTIL ANOTHER 'UNREACHABLE' MESSAGE IS RECEIVED AND THE PROCEURE REPEATS 
ITSELF
EVENTUALLY I GET -

Oct 15 23:49:52 WARNING[2490]: chan_sip.c:1208 retrans_pkt: Maximum retries 
exceeded on transmission 178ce1c47486afd264bdf26f56eb1c9e at 81.174.170.48 for 
seqno 103 (Non-critical Request)
Destroying call '178ce1c47486afd264bdf26f56eb1c9e at 81.174.170.48'


IF I DO A SIP SHOW PEERS WHILST  'JGRIFFITHS'  IS REACHABLE - I GET

asterisk*CLI> sip show peers
Name/username             Host                 Dyn     Nat     ACL     Port 
Status
jgriffiths/jgriffiths             80.176.242.9     D         N 
5060     OK (83 ms)
SPA3102/SPA3102      92.168.1.12       D                             5060 
Unmonitored
2 sip peers [2 online , 0 offline]
asterisk*CLI>

IF I DO A SIP SHOW PEERS WHILST  'JGRIFFITHS' IS UNREACHABLE - I GET

Oct 15 23:54:43 NOTICE[2490]: chan_sip.c:11328 sip_poke_noanswer: Peer 
'jgriffiths' is now UNREACHABLE! Last qualify: 115
Destroying call '6d81be4c1eeb3bcc72978a5720057279 at 81.174.170.48'
asterisk*CLI> sip show peers
Name/username         Host                     Dyn     Nat     ACL     Port 
Status
jgriffiths/jgriffiths         80.176.242.9         D        N 
5060     UNREACHABLE
SPA3102/SPA3102 192.168.1.12          D                              5060 
Unmonitored
2 sip peers [1 online , 1 offline]
asterisk*CLI>

Anyone any ideas?

Is it something to do with timing of the 'sip poke' ?  One ends timer out of 
sync with the other end?.

We are only beginners at this game but learning fast!


Ian Jolly

+44 (0)352 85 26 (via 1929 GPO UAX5 exchange)
CNET - the Heritage Telephone Network