Losing Registration with voipvoip.com

Asterisk Asterisk Support
1

Hi all
Asterisk 1.4.30
FreePbx 2.7.0.2

I am behind a standard Linksys E1000 router.

I am using a SIP connection and voipvoip.com as my SIP provider.

So i am having a problem with incoming calls.
Incoming calls usually work when I reboot the system, but after a period of time (usually about a day), the calls no longer come to the Asterisk system, but instead go to the VoipVoip standard voicemail system (indicating my Asterisk system is not connecting to the VoipVoip system)

When I run a ‘sip reload’ a few times (usually 3 times), the calls will start coming again only for about a day, then this will happen again.

Even though it seems like it is losing registration with the voipvoip server, when i run a ‘sip show registry’ - it shows it is registered with sip3.voipvoip.com:5060
Outgoing calls have never had a problem.

here is my incoming trunk configuration:

USER Context: VOIPVOIP Account Number HERE - removed number

USER Details:
username= VOIPVOIP Account Number HERE - removed number
type=user
secret=VOIPVOIP Account Password HERE - removed number
nat=auto
insecure=very
host=sip3.voipvoip.com
fromdomain=sip3.voipvoip.com
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=g729&ulaw&alaw&ilbc

Ports I have forwarded to the Asterisk system are:
UDP 10000 - 20000
UDP & TCP 5000 - 5500

In 'Asterisk Sip Settings’
I have:
NAT: no
IP Configuration: Static IP
External: MyIPAddressHere
Local Networks: 192.168.0.0 / 255.255.255.0

I changed this to NAT: yes – but then it will not register whatsoever with the voipvoip server

sip show registry shows unregistered - This is after having set NAT:yes

This has been doing this ever since the server was configured so it isn’t like it suddenly stopped working correctly.

Have you heard of this happening before and can help me fix it?

Thank you so much!! It is quite frustrating.

2

Anyone have any thoughts on this?

thank you

3

Hi forsberg94!

I got the same problem here, but with phonzo.se!
But it is a couple of days between my register drops out!

I haven’t found any solution on this anywhere but I hope somebody do have any idea on this!

4

You need at least the CLI output for the failed registration attempt, and probably sip set debug (or Wireshark) output for it.

5

Here is a part of the log from the Asterisk log file

[May 25 07:46:04] DEBUG[2861] chan_sip.c: Target address 69.90.209.56 is not local, substituting externip [May 25 07:46:04] DEBUG[2861] chan_sip.c: Scheduled a registration timeout for sip3.voipvoip.com id #21 [May 25 07:46:04] VERBOSE[2861] logger.c: REGISTER attempt 1 to 5557232074@sip3.voipvoip.com [May 25 07:46:04] DEBUG[2861] chan_sip.c: = Found Their Call ID: 6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145 Their Tag Our tag: as58c9ea24 [May 25 07:46:04] DEBUG[2861] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145' Request 105: Found [May 25 07:46:04] DEBUG[2861] chan_sip.c: = Found Their Call ID: 6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145 Their Tag Our tag: as58c9ea24 [May 25 07:46:04] DEBUG[2861] chan_sip.c: Stopping retransmission on '6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145' of Request 105: Match Found [May 25 07:46:04] DEBUG[2861] chan_sip.c: Initializing already initialized SIP dialog 6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145 (presumably reinvite) [May 25 07:46:04] VERBOSE[2861] logger.c: REGISTER attempt 2 to 5557232074@sip3.voipvoip.com [May 25 07:46:04] DEBUG[2861] chan_sip.c: = Found Their Call ID: 6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145 Their Tag Our tag: as290112a1 [May 25 07:46:04] DEBUG[2861] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145' Request 106: Found [May 25 07:46:04] DEBUG[2861] chan_sip.c: = Found Their Call ID: 6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145 Their Tag Our tag: as290112a1 [May 25 07:46:04] DEBUG[2861] chan_sip.c: Stopping retransmission on '6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145' of Request 106: Match Found [May 25 07:46:04] DEBUG[2861] chan_sip.c: Registration successful [May 25 07:46:04] DEBUG[2861] chan_sip.c: Cancelling timeout 21 [May 25 07:46:18] DEBUG[2861] chan_sip.c: = No match Their Call ID: 6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145 Their Tag 51dfe8a878c43c66d00620c1479c24ab.0c07 Our tag: as290112a1

coasterisk*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time
sip3.voipvoip.com:5060          5ACCOUNTNUMBER4         185 Registered           Tue, 25 May 2010 07:49:09
coasterisk*CLI>

I can’t exactly replicate the behavior since it is registering at times and then not at others… yet it says it is registered no matter…

I have enabled the sip debug

the only thing i can see that is relevant to the asterisk box (192.168.0.145) is this:

--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '6829c0131b7c474d54c962a64d6b6ccc@192.168.0.145' in 32000 ms (Method: REGISTER)

do i need to add more of my log? or maybe even a different log

Thanks

6

The sip debug should include the actual SIP messages sent. I think they do not come out in the debug stream, but in one of the more normal streams. I would normally uncomment the full log in logger.conf to get all the streams in one file.

The scheduling destruction message is perfectly normal.