upgraded to 2.9 and now no calls are getting connected

mrmrmrmr's picture
Posted January 30th, 2012 by mrmrmrmr (tadpole)

Hi,

I had a running FreePBX installation with lots of configuration. But it was on an old CentOS system and I decided to move to a Ubuntu server.
I got my new server ready and then installed all Asterisk packages together with FreePBX.
Free PBX 2.9 is installed. Then I took a backup from my old system with Backup &Restore module. I restored that backup to the new one.
New configuration is accepted by the FreePBX and everything seems in place.

However I can't dial out any calls.

I get "exit with zero" errors on the log. But that's all; there is no other log.

How can I analyze this further ? And any solutions ?

thanks.


__________________


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Did you restore a backup

SkykingOH's picture
On January 31st, 2012 SkykingOH (Leap Frog) said:

Did you restore a backup from a version previous to the 2.9?


yes, I did install a backup

mrmrmrmr's picture
On January 31st, 2012 mrmrmrmr (tadpole) said:

yes, I did install a backup from 2.8.
but what's wrong with that ?
configuration seems to be accepted by the system.

how can I get more logs/debugs ?


Major DB schema changes

SkykingOH's picture
On January 31st, 2012 SkykingOH (Leap Frog) said:

Major DB schema changes occurred between the two versions.

If you untar the backup file and run, move it to the tmp directory and run the generate astdb script you probably will have registrations but so much more will have to be fixed. You will find you have a ton of broken destinations.

Both DB's are completely hosed. I tried selectively restoring tables so I did not have to manually recreate 30 IVR's and it took Tony, Bryan and I almost 4 hours to fix the issues. You have to use the migration tool, period. Simply no way around it.

I know it's not the answer you wanted.


what is the migration tool

mrmrmrmr's picture
On January 31st, 2012 mrmrmrmr (tadpole) said:

what is the migration tool ?
I can use it if it's applicable.

btw, IMHO the problem isn't about the configuration because when I had initially installed my previous FreePBX setup (the working one) , I had to make changes to the dialparties.agi script to enable calls.

I just don't remember what was the modification. and as I don'thave any logs, I'm not sure if it's the same problem or not.


If you do a check for

SkykingOH's picture
On February 1st, 2012 SkykingOH (Leap Frog) said:

If you do a check for updates on a 2.9 system the "2.10" upgrade too will appear. You can't run it once the dbmigrate scripts have been run.

I can't think of any reason to modify dialparties.agi


from 2.9 to 2.10 ? how will

mrmrmrmr's picture
On February 1st, 2012 mrmrmrmr (tadpole) said:

from 2.9 to 2.10 ?
how will that help me ?

I know about the upgrade module, but that's for upgrading from a previous version to new version.

I now have 2 systems:
1. 2.8.x with a working configuration on CentOS 5
2. 2.9.x with non working configuration on Ubuntu 11.10

how can I migrate my configuration to the Ubuntu system ?

and why do I have no logging on the new system ?

thx.


so ?

mrmrmrmr's picture
On February 2nd, 2012 mrmrmrmr (tadpole) said:

so ?
any idea ?


help needed

mrmrmrmr's picture
On February 3rd, 2012 mrmrmrmr (tadpole) said:

Hi,

I'm stuck and need help.

I now have 2 systems:
1. 2.8.x with a working configuration on CentOS 5
2. 2.9.x with non working configuration on Ubuntu 11.10
how can I migrate my configuration to the Ubuntu system ?
and why do I have no logging on the new system ?

thx.


help ?

mrmrmrmr's picture
On February 7th, 2012 mrmrmrmr (tadpole) said:

Hi,

I'd appreciate any help please.

Thanks.


?

mrmrmrmr's picture
On February 12th, 2012 mrmrmrmr (tadpole) said:

SkykingOH,

are you not following up ?
I really need help on this. I'm stuck.


Following up on what?

SkykingOH's picture
On February 12th, 2012 SkykingOH (Leap Frog) said:

Following up on what?

Upgrade the 2.8 system to match the Ubuntu system.

Use the FreePBX upgrade utility. That's all you have to do.

Keep in mind 2.10 is still beta and the backup module is all new and a work in progress.

If you want follow up and accountability you need to purchase support. This is a community forum.


I don't understand

mrmrmrmr's picture
On February 15th, 2012 mrmrmrmr (tadpole) said:

Hi,

Unfortunately I don't understand:

Do you propose to upgrade the old (CentOS) system to 2.9 version first ?
And then get a backup with "Backup & restore" module ?
and import that to the new 2.9 system ?

isn't it risky to upgrade the old system to the new version ? what if I get the same problems on the functioning old system ?

btw, I know this is a community forum but you made a suggestion and I didn't understand it. That's why I'm trying to ask details...
sorry if I was not polite enough. It's just disturbing that I couldn't make a working system on the new installation with backups from old system.


It's not about being polite.

SkykingOH's picture
On February 16th, 2012 SkykingOH (Leap Frog) said:

It's not about being polite. I don't know what to tell you to do.

You are running an unsupported OS for starters.

The migration tool migrates data within the framework. You can't restore from a previous version, period.

The new system does not work because you trashed the DB with the restore and more than likely the AstDB is also in an unrestored state.

If you don't want to try the in place update then install the old version on another computer, restore to that then run the upgrade util. I have had to do that on a VM to migrate users from trixbox's.

Also if your config is not complex you can use the bulk import module, copy the sound files over SCP and then just setup everything else again by hand.


thanks.

mrmrmrmr's picture
On February 16th, 2012 mrmrmrmr (tadpole) said:

now that the new installation is trashed with old version's restore, how can I restore a fresh DB with zero config ?

how can Iremovethe current installation and install old version (2.8) ?


Depends how well you know

SkykingOH's picture
On February 17th, 2012 SkykingOH (Leap Frog) said:

Depends how well you know Linux and MySQL.

Essentially just drop asterisk and asteriskcdrdb databases from MySQL tool.

Then remove everything from the doc root /var/www/html/admin

Lastly download the 2.8 tarball, untar it, run the dbcreate scripts and then do a normal install_amp.

If this is overwhelming just go pickup an hour of support and we can do this for you.


thanks

mrmrmrmr's picture
On February 18th, 2012 mrmrmrmr (tadpole) said:

thank you. I did as you suggested (btw, I am using asterisk at home for only 2 sip channels, it's not a commercial setup.)

unfortunately after restoring my backup I have several problems. Please see below.
It seems that I have a wrong password somewhere. But I don't know where. any idea ?

1. on the console I have several of these errors:

  == Connect attempt from '127.0.0.1' unable to authenticate
  == Connect attempt from '127.0.0.1' unable to authenticate
router*CLI>

2. outgoing trunk is registered to my service provider. extensions are registered but I can't make a call.

3. on GUI, I get "asterisk connection failure"

Asterisk Manager Connection Failure

Failed to connect to the Asterisk manager through port: 5038
Added 11 minutes ago
(retrieve_conf.FATAL)

4. on GUI, I get this error:

 retrieve_conf failed, config not applied

Reload failed because retrieve_conf encountered an error: 1
Added 11 minutes ago
(freepbx.RCONFFAIL)

5. on GUI, I have brokem modules warning

You have 9 broken modules

The following modules are disabled because they are broken:
parking, paging, queues, timeconditions, languages, daynight, donotdisturb, customcontexts, vmailadmin

You should go to the module admin page to fix these.
Added 1 seconds ago
(freepbx.modules_broken)

6. on GUI, I have "Symlink from modules failed" error:

 Symlink from modules failed

retrieve_conf failed to sym link: 
   /etc/asterisk/sip_notify.conf from core/etc
   /etc/asterisk/logger.conf from core/etc
This can result in FATAL failures to your PBX. If the target file exists and not identical, the symlink will not occur and you should rename the target file to allow the automatic sym link to occur and remove this error, unless this is an intentional customization.
Added 18 minutes ago
(retrieve_conf.SYMLINK)

I think.

mrmrmrmr's picture
On February 18th, 2012 mrmrmrmr (tadpole) said:

I think I've solved the password issue by changing my password in /etc/asterisk/manager.conf
but I have the following issue which I couldn't solve:

Symlink from modules failed

retrieve_conf failed to sym link: 
   /etc/asterisk/sip_notify.conf from core/etc
   /etc/asterisk/logger.conf from core/etc
This can result in FATAL failures to your PBX. If the target file exists and not identical, the symlink will not occur and you should rename the target file to allow the automatic sym link to occur and remove this error, unless this is an intentional customization.
Added 25 seconds ago
(retrieve_conf.SYMLINK)

and my calls are still not connecting.

btw, I still have broken modules. I didn't want to fix them before anyone makes a suggestion about above issues...


ok

mrmrmrmr's picture
On February 18th, 2012 mrmrmrmr (tadpole) said:

OK; I've also solved the symlink issue.
I removed the /etc/asterisk/logger.conf and /etc/asterisk/sip_notify.conf
now I don't have this issue.

also I have manually installed broken modules. now I don't have any errors on the GUI and console.

but my calls are still not connecting.

please help me...


You are not telling us much

SkykingOH's picture
On February 18th, 2012 SkykingOH (Leap Frog) said:

You are not telling us much (output of sip show peers, debug info) I have no idea what "calls don't work" means exactly.

What I suspect is the astdb is not restored properly. I mentioned this in the info I provided you two weeks ago.

Quote:
If you untar the backup file and run, move it to the tmp directory and run the generate astdb script you probably will have registrations but so much more will have to be fixed. You will find you have a ton of broken destinations.


ok

mrmrmrmr's picture
On February 18th, 2012 mrmrmrmr (tadpole) said:

"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...


Nobody tries to help because

SkykingOH's picture
On February 19th, 2012 SkykingOH (Leap Frog) said:

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.


wait

mrmrmrmr's picture
On February 19th, 2012 mrmrmrmr (tadpole) said:

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.


Wow, huge statements that

SkykingOH's picture
On February 20th, 2012 SkykingOH (Leap Frog) said:

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.


answers

mrmrmrmr's picture
On February 20th, 2012 mrmrmrmr (tadpole) said:

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

1. so you recommend doing a restore by running the php script.

2. 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.

3. next time , I'll consider CentOS. I had thought that Ubuntu is a well known distro and would be supported anyway.

4. 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...

5. I'll try those debug commands.


You have a ton of Linux

SkykingOH's picture
On February 21st, 2012 SkykingOH (Leap Frog) said:

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.


logs and show outpur

mrmrmrmr's picture
On February 26th, 2012 mrmrmrmr (tadpole) said:

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 <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': 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:02165563000@192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-323df9e4
From: Home <sip:995@192.168.254.254>;tag=751feec6ea016d94o0
To: <sip:02165563000@192.168.254.254>
Call-ID: db5d042e-8215174c@192.168.254.5
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Home <sip:995@192.168.254.5:5060>
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 - db5d042e-8215174c@192.168.254.5
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:995@192.168.254.254>;tag=751feec6ea016d94o0
To: <sip:02165563000@192.168.254.254>;tag=as768ce7ca
Call-ID: db5d042e-8215174c@192.168.254.5
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 'db5d042e-8215174c@192.168.254.5' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.254.5:5060 --->
ACK sip:02165563000@192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-323df9e4
From: Home <sip:995@192.168.254.254>;tag=751feec6ea016d94o0
To: <sip:02165563000@192.168.254.254>;tag=as768ce7ca
all-ID: db5d042e-8215174c@192.168.254.5
CSeq: 101 ACK
Max-Forwards: 70
Contact: Home <sip:995@192.168.254.5:5060>
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:02165563000@192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-88d01be0
From: Home <sip:995@192.168.254.254>;tag=751feec6ea016d94o0
To: <sip:02165563000@192.168.254.254>
Call-ID: db5d042e-8215174c@192.168.254.5
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="995",realm="asterisk",nonce="39a67bbc",uri="sip:02165563000@192.168.254.254",algorithm=MD5,response="7dfadfe9c5508258626d29b4fd34682c"
Contact: Home <sip:995@192.168.254.5:5060>
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 - db5d042e-8215174c@192.168.254.5
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:995@192.168.254.254>;tag=751feec6ea016d94o0
To: <sip:02165563000@192.168.254.254>;tag=as768ce7ca
all-ID: db5d042e-8215174c@192.168.254.5
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 'db5d042e-8215174c@192.168.254.5' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.254.5:5060 --->
ACK sip:02165563000@192.168.254.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.5:5060;branch=z9hG4bK-88d01be0
From: Home <sip:995@192.168.254.254>;tag=751feec6ea016d94o0
To: <sip:02165563000@192.168.254.254>;tag=as768ce7ca
Call-ID: db5d042e-8215174c@192.168.254.5
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="995",realm="asterisk",nonce="39a67bbc",uri="sip:02165563000@192.168.254.254",algorithm=MD5,response="7dfadfe9c6d08258626d29bb4d34682c"
Contact: Home <sip:995@192.168.254.5:5060>
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0

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

anyone understands the problem ffrom these logs ?

mrmrmrmr's picture
On March 10th, 2012 mrmrmrmr (tadpole) said:

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


clean install

mrmrmrmr's picture
On March 18th, 2012 mrmrmrmr (tadpole) said:

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-ins...


giving up

mrmrmrmr's picture
On April 1st, 2012 mrmrmrmr (tadpole) said:

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)