Help!!!!!!!! TRUNKS not registering?

netizen

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#1
I don't know what is happening really...please help as I need to get those sip calls back! :(
The problem:
The Trunks are not registering. "sip show registry" shows nothing.
The extensions though are ok and the calls between them work just fine.
Where should I look?

I haven't changed anything unless the update I've done a few days back broke something and I am only seeing it today! :(

Any help is much appreciated!!!!

Thank you!
 

netizen

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#2
Any ideas anyone?
I've been trying to find out what's going on and I can't figure it out. :(
All trunks (various sip trunk providers) do not register. There is NOTHING in the logs to show even any failed attempts. It is like Elastix is not trying to login to remote providers.

Also, I have several incoming SIP numbers which are routed to my IP directly without the need of a registration...even this one is not working.
Strange enough however all my sip extensions are working fine.

I have access to another server so I tried to create a trunk from my problematic server to an extensions of the second server. Again nothing. No trace on logs of both machines of an attempt to register! The trunk appears as normal in FreePBX.

SIP show registry still shows nothing.
I'm puzzled :(((((((((((

Heeeeelp! :(((

N


netizen said:
I don't know what is happening really...please help as I need to get those sip calls back! :(
The problem:
The Trunks are not registering. "sip show registry" shows nothing.
The extensions though are ok and the calls between them work just fine.
Where should I look?

I haven't changed anything unless the update I've done a few days back broke something and I am only seeing it today! :(

Any help is much appreciated!!!!

Thank you!
 

Patrick_elx

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#3
did you try to reboot your server?
Are you looking at the message on the screen during the boot to see if there are any errors?
Did you try to reboot your router/DNS?

What is your setup?

What kind of trunk are you using?

With the few information you gave us it's hard to help.
 

netizen

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#4
Hello and thank you for the reply.
I wouldn't call myself a newbie but neither an Elastix guru...
I've done most of the basics (apart from the backup!)

Some answers to your questions:
-I did try to reboot the server a few times.
-I cannot look at the screen during bootup simply because the server is in a data center.
-I am only using SIP trunks (i.e no cards installed - never were)
-The server also acts as a hosting server for me and friends. All applications (WEB,FTP,SSH,MySQL,Postfix -mail-, SpamAssasin etc) are working 100% fine.
-The server has software firewall which I disabled to troubleshoot the problem. No difference. No registration activity in the logs. I am watching the logs live from SSH console using the command tail -f /var/log/asterisk/full. When remote extensions connect to my server I can see it AND they are working fine.
-As I mentioned before, one provider sends directly a DID to me without having a trunk setup in freepbx. This used to work fine. Now, that DID does not work and when I call the number from a landline, AGAIN there is no activity in the logs!

As you can imagine I cannot request a router reboot from the data center. I don't see the reason in doing so anyway since the extensions connect normaly on port 5060!

Any ideas? :(
 

dicko

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#5
It sounds like your VSP is broke,

try

tcpdump host <ip of VSP>

for low level traffic either way.

sip debug ip <ip of VSP>

in Asterisk for higher level traffic.
 

netizen

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#6
Hello there.
I assume that by VSP you mean my provider.
Just a reminder that I have multiple providers. None works for incoming calls.
Below is output from the commands you mentioned. NOTE: when I connect to the web interface of that provider I see the following amongst other things:
Last time you account registered was: 2010-01-20 18:42:25.0
IP Address: (my server's static IP).

The above make me think that this particular provider still remembers my IP (and hence the output bellow)
I have another trunk from a BETAMAX brand which includes a DID. When I dial that DID from a landline, Betamax reports :"This user is currently offline".

There is definitely not problem with the providers.... It's definitely in my server :(
Output follows:
---------------

Command: tcpdump host <provider's IP> (while server is idle. Attempting an incoming call makes no difference)

21:26:35.583215 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:26:35.685169 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:26:49.510241 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:26:49.612082 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:27:35.684921 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:27:35.786820 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:27:49.611700 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:27:49.713647 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:28:35.786546 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:28:35.888234 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:28:49.713850 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:28:49.815637 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:29:35.887819 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:29:35.989593 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:29:49.815843 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:29:49.917691 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:30:35.989487 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:30:36.091403 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:30:49.918146 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:30:50.020063 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:31:36.091833 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:31:36.193587 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455
21:31:50.020934 IP my-hostname.sip > provider's-hostname.sip: SIP, length: 491
21:31:50.122747 IP provider's-hostname.sip > my-hostname.sip: SIP, length: 455

Command as above but when an outgoing call is attempted (***and works ok!***)
22:02:23.804931 IP my-hostname.18586 > provider's-hostname.19742: UDP, length 172
22:02:23.818750 IP provider's-hostname.19742 > my-hostname.18586: UDP, length 172
22:02:23.824592 IP my-hostname.18586 > provider's-hostname.19742: UDP, length 172
22:02:23.839505 IP provider's-hostname.19742 > my-hostname.18586: UDP, length 172
22:02:23.845311 IP my-hostname.18586 > provider's-hostname.19742: UDP, length 172
22:02:23.859054 IP provider's-hostname.19742 > my-hostname.18586: UDP, length 172
NOTE: right after trying the above command I checked the web interface at the provider and the last registration date was still the same! (2010-01-20 18:42:25.0)
Clearly my server is not registering any trunks.....

-----------------------------------------

Command: sip debug ip <provider's IP>
( I replaced IPs with 1.1.1.1 as my IP and 2.2.2.2 as my provider's IP )

[root@my-server ~]# asterisk -rvvvvvvvvvvvvvvvvvv
Asterisk 1.4.28, Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.28 currently running on my-server (pid = 5326)
Verbosity is at least 18
my-server*CLI> sip debug ip 2.2.2.2
SIP Debugging Enabled for IP: 2.2.2.2
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.

my-server*CLI> sip set debug ip 2.2.2.2
SIP Debugging Enabled for IP: 2.2.2.2
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
OPTIONS sip:2.2.2.2 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4b1b372a;rport
From: "Unknown" <sip:Unknown@1.1.1.1>;tag=as199832b2
To: <sip:2.2.2.2>
Contact: <sip:Unknown@1.1.1.1>
Call-ID: 5d069db247b7a45607cb64bd4e2a66ee@1.1.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 28 Jan 2010 21:33:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0



---
my-server*CLI>
<--- SIP read from 2.2.2.2:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4b1b372a;received=1.1.1.1;rport=5060
From: "Unknown" <sip:Unknown@1.1.1.1>;tag=as199832b2
To: <sip:2.2.2.2>;tag=as3928d581
Call-ID: 5d069db247b7a45607cb64bd4e2a66ee@1.1.1.1
CSeq: 102 OPTIONS
Server: MyProvider
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '5d069db247b7a45607cb64bd4e2a66ee@1.1.1.1' Method: OPTIONS
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
OPTIONS sip:2.2.2.2 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6cb16ac7;rport
From: "Unknown" <sip:Unknown@1.1.1.1>;tag=as42a33764
To: <sip:2.2.2.2>
Contact: <sip:Unknown@1.1.1.1>
Call-ID: 54d67f9929c342d06c6745ea19256390@1.1.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 28 Jan 2010 21:33:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0



---
my-server*CLI>
<--- SIP read from 2.2.2.2:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6cb16ac7;received=1.1.1.1;rport=5060
From: "Unknown" <sip:Unknown@1.1.1.1>;tag=as42a33764
To: <sip:2.2.2.2>;tag=as0e6a2409
Call-ID: 54d67f9929c342d06c6745ea19256390@1.1.1.1
CSeq: 102 OPTIONS
Server: MyProvider
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '54d67f9929c342d06c6745ea19256390@1.1.1.1' Method: OPTIONS
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
OPTIONS sip:2.2.2.2 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK25a24d78;rport
From: "Unknown" <sip:Unknown@1.1.1.1>;tag=as5faca60e
To: <sip:2.2.2.2>
Contact: <sip:Unknown@1.1.1.1>
Call-ID: 68623b9d0190a3270c9cb59265cbd0c8@1.1.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 28 Jan 2010 21:34:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0



---
Retransmitting #1 (no NAT) to 2.2.2.2:5060:
OPTIONS sip:2.2.2.2 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK25a24d78;rport
From: "Unknown" <sip:Unknown@1.1.1.1>;tag=as5faca60e
To: <sip:2.2.2.2>
Contact: <sip:Unknown@1.1.1.1>
Call-ID: 68623b9d0190a3270c9cb59265cbd0c8@1.1.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 28 Jan 2010 21:34:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
my-server*CLI>
<--- SIP read from 2.2.2.2:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK25a24d78;received=1.1.1.1;rport=5060
From: "Unknown" <sip:Unknown@1.1.1.1>;tag=as5faca60e
To: <sip:2.2.2.2>;tag=as5ac67ad9
Call-ID: 68623b9d0190a3270c9cb59265cbd0c8@1.1.1.1
CSeq: 102 OPTIONS
Server: MyProvider
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0



<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '68623b9d0190a3270c9cb59265cbd0c8@1.1.1.1' Method: OPTIONS
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
OPTIONS sip:2.2.2.2 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK0f53ce0a;rport
From: "Unknown" <sip:Unknown@1.1.1.1>;tag=as69113e24
To: <sip:2.2.2.2>
Contact: <sip:Unknown@1.1.1.1>
Call-ID: 7f605307558500c7463051971d55b87d@1.1.1.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 28 Jan 2010 21:34:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
my-server*CLI>
<--- SIP read from 2.2.2.2:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK0f53ce0a;received=1.1.1.1;rport=5060
From: "Unknown" <sip:Unknown@1.1.1.1>;tag=as69113e24
To: <sip:2.2.2.2>;tag=as0f087077
Call-ID: 7f605307558500c7463051971d55b87d@1.1.1.1
CSeq: 102 OPTIONS
Server: MyProvider
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '7f605307558500c7463051971d55b87d@1.1.1.1' Method: OPTIONS
-- Remote UNIX connection
-- Remote UNIX connection disconnected


----------------------------------------------------


Any obvious errors?
 

blangys

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#7
I'm not sure how you would get the sip trunk to work without building it in FreePBX - I assume that you built it manually. Why not comment out your trunk setup, and build it again in the GUI to test it out? Unless you are dealing with a large number of DID numbers, it seems like it would be a good test.

It's strange that the debug doesn't show the registration attempts. I've had all kinds of SIP trunk issues and all of them have been right behind my keyboard. Each time, however, the system showed the registration attempt at regular intervals.

Restore to your last known good backup.

Quickly setup another server and test out your trunk settings from another location.

Good Luck.
 

netizen

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#8
Hi and thank you for your reply.
I don't know if I gave the impression that I'm not using FreePBX.
I have deleted-recreated (the trunk) & rebooted the server several times... :(

Trunks are working fine from a test server (tested before posting in the forums...).
:(

Regarding full restore I will try another route first before giving a few hundred dollars to the datacenter for "remote hands"....

In the mean time if someone has a suggestion please let me know.
Thank you
 

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