Upgraded to 2.9 and now no calls are getting connected

mrmrmrmr · February 18, 2012, 11:02pm

“calls don’t work” means:

incoming calls (from trunk) are not ringing on my registered peers (extensions)
extensions can’t dial out
2 extension can’t call each other

so, any basic call scenario fails.

what does “astdb is not restored properly” mean ? how can I fix that ?

what do you mean by “untar the backup file and run” ?
there is nothing in that tarfile to run. I restore it through “backup & restore” module.

where is the “generate astdb” script ?

I’m sorry to disturb you by my noob questions but you are my only hope as noone else tries to help…

btw, “sip show peers” output is exactly as in my working old system. peers are registered. I don’t know what “debug info” is. if you tell me how to get that, I’ll try…

SkykingOH · February 19, 2012, 2:35am

Nobody tries to help because you are using an unsupported OS and did not follow advice to not restore between versions.

It should have been to untar the backup to the /tmp directory, then run the /var/lib/asterisk/bin/restoreastdb.php script. You will have to look at the script and see exactly what the argument it is looking for, I think it is the backup sequence number.

As far as Asterisk debug info, how to log, post logs, sip debug has been discussed on every Asterisk forum for years. I quick search of google or reading the Asterisk documentation would help you understand what is going on and what we need to help you.

I also suggested that using our support would get this resolved very quickly.

mrmrmrmr · February 19, 2012, 9:41pm

really ?

Ubuntu is an unsupported OS ?
but I am following the advice. You told me to install the older version (2.8 in my case) and I did it.
But this method didn’t help me…

So now you recommend me to untar the backup and run the restoreastdb.php script.
I’ll do that…
But I doubt it will help, because I believe that this script is automatically run by the “backup & restore module” when it applies a restore.

you also suggest to get a commercial support. but I told you that this is not a commercial setup. I use it at home as a 2 channel home PBX.
I don’t have the money for any commercial support.

p.s: I really doubt, if I had the money for commercial support, that my problem would be solved easily. There is a problem which breaks all calls on the platform. There is no clue of the root cause.
This thing has no meaningful logs…

p.s-2: I still don’t see how to get debug info that you’re looking for. no google search could answer about this.

SkykingOH · February 20, 2012, 1:03am

Wow, huge statements that are completely inaccurate:

1 - the astdb restore is run, it just doesn’t always work. Do a ‘database show’ and take a look at what keys are in that DB.

2 - I am sure I could fix it in under 30 minutes as could any member of the support team

3 - CentOS id the only Linux flavor that is supported. Everything is is “you are on your own”.

4 - You rolled back your FreePBX to 2.8, did the restore and you still can’t register?

5 - No meaningful logs? Have you even looked at Asterisk? SIP transaction logging (sip set debug peer xxx), dialplan logging (verbosity) and debug (core set debug xxx) produce more logging that you could possibly digest.

mrmrmrmr · February 20, 2012, 8:21pm

well, I didn’t know that Ubuntu is not supported. sorry…

so you recommend doing a restore by running the php script.
if you believe that this is an easy problem that can be solved in under 30 minutes, just let me know where to look at. I am using Linux distros for more than 15 years and have decent amount of linux knowledge. if it’s so easy, I could fix it by your instructions.
next time , I’ll consider CentOS. I had thought that Ubuntu is a well known distro and would be supported anyway.
I told you in previous posts that I followed your recommendation and installed a fresh 2.8 version. Then I restored the backup through “backup & restore” module.
Calls are still not being connected.
I never said that I am unable to register. My phones at home are registered and the outbound trunk is registered to my service provider sip proxy.
It’s just a matter of calls not being connected.
On my previous installation, I had “custom context” module installed. I suspected that and removed it. Still not working…
I’ll try those debug commands.

SkykingOH · February 21, 2012, 12:10am

You have a ton of Linux knowledge but you don’t know much about Asterisk or FreePBX. We could play 20 questions for hours and I may not stumble upon what you did.

Now you are making me wonder what “calls not being connected means”. The log from when a call fails is helpful.

mrmrmrmr · February 26, 2012, 5:48pm

hi,

today I tried to get the logs but I see no matter which verbosity is set I don’t get any logs except sip.
please see below, maybe you can find something meaningful. I don’t…

root@router:~# asterisk -rvvvvvvvvvvvvvvvvvv -ddddddddddddddddddddddddd
Asterisk 1.8.4.4~dfsg-2ubuntu1, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
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': Parsing /etc/asterisk/asterisk.conf
  == Found
Seeding global EID '00:0d:b9:12:cf:90' from 'eth0' using 'siocgifhwaddr'
  == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf
  == Found
Connected to Asterisk 1.8.4.4~dfsg-2ubuntu1 currently running on router (pid = 3527)
Verbosity is at least 18
Core debug is at least 25
router*CLI> sip show peers
Name/username              Host                                    Dyn Forcerport ACL Port     Status     
901/901                    46.2.136.4                               D   N   A  5060     OK (104 ms) 
902                        (Unspecified)                            D   N   A  0        UNKNOWN    
903                        (Unspecified)                            D   N   A  0        UNKNOWN    
905                        (Unspecified)                            D   N   A  0        UNKNOWN    
906                        (Unspecified)                            D   N   A  0        UNKNOWN    
990                        (Unspecified)                            D   N   A  0        UNKNOWN    
993                        (Unspecified)                            D       A  0        UNKNOWN    
995/995                    192.168.254.5                            D       A  5060     OK (12 ms) 
996                        (Unspecified)                            D   N   A  0        UNKNOWN    
998                        (Unspecified)                            D       A  0        UNKNOWN    
999/999                    192.168.254.11                           D       A  5060     OK (8 ms)  
aktun-istanbul/200004908  193.243.202.97                                      5060     Unmonitored 
aktunx-istanbul/20018090  193.243.202.97                                      5060     Unmonitored 
pstn-sip/PSTN              192.168.254.5                                       5061     Unmonitored 
14 sip peers [Monitored: 3 online, 8 offline Unmonitored: 3 online, 0 offline]
router*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time                 
193.243.202.97:5060                     N      200004908502       585 Registered           Sun, 26 Feb 2012 19:41:06
193.243.202.97:5060                     N      200180908502       585 Registered           Sun, 26 Feb 2012 19:41:06
2 SIP registrations.
router*CLI> 
router*CLI> sip set debug ip 193.243.202.97
SIP Debugging Enabled for IP: 193.243.202.97
router*CLI> sip set debug peer 995
SIP Debugging Enabled for IP: 192.168.254.5

<--- SIP read from UDP:192.168.254.5:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-323df9e4
From: Home <sip:[email protected]>;tag=751feec6ea016d94o0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Home <sip:[email protected]:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 310
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 51628835 51628835 IN IP4 192.168.254.5
s=-
c=IN IP4 192.168.254.5
t=0 0
m=audio 16450 RTP/AVP 18 0 8 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
Sending to 192.168.254.5:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '995' for '995' from 192.168.254.5:5060

<--- Reliably Transmitting (no NAT) to 192.168.254.5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-323df9e4;received=192.168.254.5
From: Home <sip:[email protected]>;tag=751feec6ea016d94o0
To: <sip:[email protected]>;tag=as768ce7ca
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-2.8.1(1.8.4.4)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39a67bbc"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.254.5:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-323df9e4
From: Home <sip:[email protected]>;tag=751feec6ea016d94o0
To: <sip:[email protected]>;tag=as768ce7ca
all-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: Home <sip:[email protected]:5060>
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.254.5:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-88d01be0
From: Home <sip:[email protected]>;tag=751feec6ea016d94o0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="995",realm="asterisk",nonce="39a67bbc",uri="sip:[email protected]",algorithm=MD5,response="7dfadfe9c5508258626d29b4fd34682c"
Contact: Home <sip:[email protected]:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 310
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 51628835 51628835 IN IP4 192.168.254.5
s=-
c=IN IP4 192.168.254.5
t=0 0
m=audio 16450 RTP/AVP 18 0 8 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.254.5:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '995' for '995' from 192.168.254.5:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.254.5:16450
Looking for 02165563000 in access (domain 192.168.254.254)

<--- Reliably Transmitting (no NAT) to 192.168.254.5:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-88d01be0;received=192.168.254.5
From: Home <sip:[email protected]>;tag=751feec6ea016d94o0
To: <sip:[email protected]>;tag=as768ce7ca
all-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.4.4)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.254.5:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-88d01be0
From: Home <sip:[email protected]>;tag=751feec6ea016d94o0
To: <sip:[email protected]>;tag=as768ce7ca
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="995",realm="asterisk",nonce="39a67bbc",uri="sip:[email protected]",algorithm=MD5,response="7dfadfe9c6d08258626d29bb4d34682c"
Contact: Home <sip:[email protected]:5060>
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
router*CLI>

mrmrmrmr · March 10, 2012, 10:30pm

so, I just wonder…
has anyone understood the problem from these logs ?

mrmrmrmr · March 18, 2012, 3:45pm

Hi,

As I couldn’t make it run with my old configuration, I’m now trying with a clean install.
I installed version 2.9.0 and created 1 extension with my outbound routes and trunks.
Unfortunately outbound call still does not work.
My phone receives a “DECLINED” message from Asterisk and I don’t understand why.

I created a new thread for this as it’s a new installation:

http://www.freepbx.org/forum/freepbx/installation/failure-with-clean-installation-receive-603-declined-from-pbx

mrmrmrmr · April 1, 2012, 5:01pm

tried everything. including installation of a new system with a very simple configuration.
Asterisk and FreePBX does not send any packets to the trunk.
giving up because this is not working and there is no one else on the forum willing to help. (except SkykingOH whom I’d like to thank although he couldn’t show me the correct path)