Hi guys,
I have a problem with my Trixbox and I can't seem to figure out why. I believe lots of people have come across this and I'm sure the solution is posted on here somewhere but I'm hopeless to find one that fixes my problem
I have a sip trunk with peer settings as follows:
fromdomain=sip.gotalk.com
fromuser=09XX
host=sip.gotalk.com
insecure=very
nat=yes
port=5060
qualify=yes
secret=XXX
type=peer
username=09XX
and user settings:
context=from-trunk
fromdomain=sip.gotalk.com
host=sip.gotalk.com
secret=XXX
type=user
username=09XX
with register string: 09XX:XXX@sip.gotalk.com/09XX
This is all working fine. However I'm on dynamic IP (my ISP seems to enjoy switching it every couple of days), and everytime it's changed, the sip peer's status appears to be UNREACHABLE. From the log, i've noticed that the registration kept timing out and being resent many times. Here's part of my log:
---------------------------------------------
Sep 26 11:05:52 NOTICE[15678] chan_sip.c: -- Registration for '09437254@sip.gotalk.com' timed out, trying again (Attempt #44)
Sep 26 11:05:52 DEBUG[15678] chan_sip.c: Stopping retransmission on '6af227be56b810692f0dde372fef59b2@127.0.0.1' of Request 145: Match Found
Sep 26 11:05:52 DEBUG[15678] acl.c: ##### Testing 202.169.178.10 with 10.1.2.0
Sep 26 11:05:52 DEBUG[15678] chan_sip.c: Target address 202.169.178.10 is not local, substituting externip
Sep 26 11:05:52 DEBUG[15678] chan_sip.c: Scheduled a registration timeout for sip.gotalk.com id #1408
Sep 26 11:05:53 DEBUG[15678] acl.c: ##### Testing 202.169.178.10 with 10.1.2.0
Sep 26 11:05:53 DEBUG[15678] chan_sip.c: Target address 202.169.178.10 is not local, substituting externip
Sep 26 11:05:53 DEBUG[15678] acl.c: ##### Testing 10.1.2.103 with 10.1.2.0
Sep 26 11:05:53 DEBUG[15678] chan_sip.c: Stopping retransmission on '41727e4d741bf164408bdd0717a833b2@10.1.2.10' of Request 102: Match Found
---------------------------------------------
And here's the sip debug from the CLI:
---------------------------------------------
---
asterisk*CLI>
Retransmitting #2 (no NAT) to 202.169.178.10:5060:
REGISTER sip:sip.gotalk.com SIP/2.0
Via: SIP/2.0/UDP 219.90.175.100:5060;branch=z9hG4bK2ac18436;rport
From:
To:
Call-ID: 6af227be56b810692f0dde372fef59b2@127.0.0.1
CSeq: 151 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact:
Event: registration
Content-Length: 0
---
asterisk*CLI>
Retransmitting #4 (NAT) to 202.169.178.10:5060:
OPTIONS sip:sip.gotalk.com SIP/2.0
Via: SIP/2.0/UDP 219.90.175.100:5060;branch=z9hG4bK0d9810b4;rport
From: "Unknown"
To:
Contact:
Call-ID: 0d985ff551b53b754da77a1956048a19@219.90.175.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 26 Sep 2007 01:37:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '0d985ff551b53b754da77a1956048a19@219.90.175.100'
Retransmitting #3 (no NAT) to 202.169.178.10:5060:
REGISTER sip:sip.gotalk.com SIP/2.0
Via: SIP/2.0/UDP 219.90.175.100:5060;branch=z9hG4bK2ac18436;rport
From:
To:
Call-ID: 6af227be56b810692f0dde372fef59b2@127.0.0.1
CSeq: 151 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact:
Event: registration
Content-Length: 0
---
Retransmitting #4 (no NAT) to 202.169.178.10:5060:
REGISTER sip:sip.gotalk.com SIP/2.0
Via: SIP/2.0/UDP 219.90.175.100:5060;branch=z9hG4bK2ac18436;rport
From:
To:
Call-ID: 6af227be56b810692f0dde372fef59b2@127.0.0.1
CSeq: 151 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact:
Event: registration
Content-Length: 0
---
Retransmitting #5 (no NAT) to 202.169.178.10:5060:
REGISTER sip:sip.gotalk.com SIP/2.0
Via: SIP/2.0/UDP 219.90.175.100:5060;branch=z9hG4bK2ac18436;rport
From:
To:
Call-ID: 6af227be56b810692f0dde372fef59b2@127.0.0.1
CSeq: 151 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact:
Event: registration
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (NAT) to 202.169.178.10:5060:
OPTIONS sip:sip.gotalk.com SIP/2.0
Via: SIP/2.0/UDP 219.90.175.100:5060;branch=z9hG4bK5604ac55;rport
From: "Unknown"
To:
Contact:
Call-ID: 57bde5cf363875e014fd30be050ba2ab@219.90.175.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 26 Sep 2007 01:37:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Retransmitting #1 (NAT) to 202.169.178.10:5060:
OPTIONS sip:sip.gotalk.com SIP/2.0
Via: SIP/2.0/UDP 219.90.175.100:5060;branch=z9hG4bK5604ac55;rport
From: "Unknown"
To:
Contact:
Call-ID: 57bde5cf363875e014fd30be050ba2ab@219.90.175.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 26 Sep 2007 01:37:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Retransmitting #2 (NAT) to 202.169.178.10:5060:
OPTIONS sip:sip.gotalk.com SIP/2.0
Via: SIP/2.0/UDP 219.90.175.100:5060;branch=z9hG4bK5604ac55;rport
From: "Unknown"
To:
Contact:
Call-ID: 57bde5cf363875e014fd30be050ba2ab@219.90.175.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 26 Sep 2007 01:37:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---------------------------------------------
When IP changes and i discover quick enough to do a "reload" it's fine. However, if it happens overnight, sip trunk becomes UNREACHABLE. A reboot of the machine will normally fix the problem.
Your help is very much appreciated.
Thanks
Tho
Member Since:
2007-04-10