inbound calls don't work after update 1.1.6 to 1.3

unikbit

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#1
Hello,

After update from 1.16 to 1.3 when I try to call my elastix tone I get a silent and than a busy tone.

Thanks in advance.
Silviu
 

centritech

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#2

unikbit

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#3
Hello,

I have tryed
cp /etc/asterisk/zapata.conf.rpmnew /etc/asterisk/zapata.conf
amportal restart

but still don't work

thanks
 

unikbit

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#4
I have to specify the line is a SIP one, and I can see my line in FOP and also in Free PBX look active line.
 

jgutierrez

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#5
one question if you executes:
asterisk -rx "sip show peers"

what is the status of your sip trunk?
don't forget to paste the output
 

unikbit

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#6
Hi,

Get this:

[root@host ~]# asterisk -rx "sip show peers"
Name/username Host Dyn Nat ACL Port Status
terrasip/21100017918 193.47.84.4 5060 Unmonitored
alonou/40332560285 193.16.148.244 5060 Unmonitored
3001/3001 (Unspecified) D N 0 UNKNOWN
3000/3000 (Unspecified) D N 0 UNKNOWN
4 sip peers [Monitored: 0 online, 2 offline Unmonitored: 2 online, 0 offline]

Thanks
 

jgutierrez

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#7
mmm...

I supose that when you try to dial out, extension 3000 or 3001 is registered, isn't?

Please check the Unembedded freePBX section, and see if you have any Free PBX Notices, copy and paste them in here, also the details from each of it
 

unikbit

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#8
To dial out is working, and also when call extensions 3000 or 3001. I have attached an image with system status of FreePBX

Free PBX Notice

*
Critical Error You have a broken module
Ignore this
The following modules are disabled because they are broken:
asterisk-cli

You should go to the module admin page to fix these.
Added ago
(freepbx.modules_broken)
*
Update There is 1 module available for online upgrade
Ignore this
cidlookup 2.5.0.4 (current: 2.5.0.3)

Added 14 minutes ago
(freepbx.NEWUPDATES)
*
Error There are 1 bad destinations
Ignore this
DEST STATUS: EMPTY
Queue: AGENT2 (345)

Added 20 hours, 28 minutes ago
(retrieve_conf.BADDEST)
*
Notice Memory Limit Changed
Delete thisIgnore this
Your memory_limit, 16M, is set too low and has been increased to 100M. You may want to change this in you php.ini config file
Added ago
(core.MEMLIMIT)
*
Notice 53 New modules are available
Delete thisIgnore this
The following new modules are available for download. Click delete icon on the right to remove this notice.
findmefollow (2.5.1.6)
dashboard (2.5.0.1)
ringgroups (2.5.1.6)
featurecodeadmin (2.5.0.2)
music (2.5.1)
irc (2.5.0)
queueprio (2.5.0.4)
phpagiconf (2.5.0.2)
blacklist (2.5.0.2)
disa (2.5.1.5)
speeddial (2.5.0)
fw_ari (2.5.1.1)
queues (2.5.4.4)
phpinfo (2.4.0)
javassh (2.5.0)
languages (2.5.0.6)
dictate (2.5.0.1)
cidlookup (2.5.0.3)
pbdirectory (2.4.0.2)
fw_fop (2.5.0)
customerdb (2.5.0.3)
phonebook (2.5.0.2)
vmblast (2.5.0.2)
parking (2.5.1.2)
callforward (2.5.0.1)
backup (2.5.1.4)
paging (2.5.0.6)
pinsets (2.5.0.1)
miscapps (2.5.0.2)
callback (2.5.0.2)
voicemail (2.5.1.3)
recordings (3.3.8.8)
miscdests (2.5.0.1)
dundicheck (2.5.0)
asteriskinfo (2.5.0.1)
daynight (2.5.0.8)
fw_langpacks (2.5.0.2)
core (2.5.0.2)
timeconditions (2.5.0.6)
donotdisturb (2.5.0.4)
conferences (2.5.1.4)
announcement (2.5.1.6)
logfiles (2.5.0)
ivr (2.5.20.4)
asterisk-cli (2.5.0.2)
inventorydb (2.5.0.1)
printextensions (2.5.0.2)
manager (2.5.0.1)
customappsreg (2.5.0.3)
infoservices (2.5.0.1)
gabcast (2.5.0)
callwaiting (2.4.0)
framework (2.5.0.1)

Added 3 days, 20 hours, 33 minutes ago
(freepbx.NEWMODS)

show all
 

jgutierrez

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#9
ok,
please run the following:
asterisk -rx "zap show channels"
What do you get?
 

unikbit

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#10
Got this:

[root@host ~]# asterisk -rx "zap show channels"
Chan Extension Context Language MOH Interpret
pseudo default default
The 'zap show channels' command is deprecated and will be removed in a future re lease. Please use 'dahdi show channels' instead.
[root@host ~]# asterisk -rx "dahdi show channels"
Chan Extension Context Language MOH Interpret
pseudo default default
[root@host ~]#

Maybe help: I have only one SIP account for inbound:
alonou/40332560285 193.16.148.244 5060 Unmonitored
The inbound route goes to IVR so not to an extention. Before the update when I called 0040332560285 I heard my audio file.

Thanks,


Thanks
 

jgutierrez

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#11
mmmm...

If you want, you may contact me at
  • jgutierr_007@hotmail.com
so I may login into your server
 

unikbit

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#12
ok, I have sent therver details.

Thanks a lot
 

unikbit

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#13
any other ideas?
I couldn't take contact with jgutierrez or he is too busy, so the problem is not solved
thanks
 

damarist

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#14
Re:inbound calls don't work after update 1.1.6 to

I had the same problem.

I started with Elastix 1.1.8 in a server and then I was updating the server continuously.
The server has a TDM800P Digium card and everything was working fine, since May 2,008.

The last Friday (Oct. 17th.) I updated the server again, and then the calls trough the trunks failed (the internal calls sip-sip sip-iax did not have this symptom).
The error was:

"Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)"

I was looking for a solution a lot of time and I can't find anything to repair the problem yet :S

But I have another Elastix server with TDM400P Digium Card. This server started with Elastix 1.2.4 and I was updated this server too, and everything works fine.

The difference is the computer hardware!
I think that the problem could be something about it, but I'm not sure :( because:

I made a server backup, and tried to install Elastix version 1.3 from de iso image (http://sourceforge.net/project/showfile ... _id=629777)
in another server with the same card (TDM800P), but the problem appeared again :dry:

Then, in the new server today installed Elastix version 1.2.4 (I don't now if this version still could be in the sourcforge project :( ) but I configured everything again, because I tried with a backup (of the server with the problem) to restore the configurations, but I don't now if at the end by the stress I culdn't to restore the data :(

Now, I have the 1.2.4 working and I don't to try to do an update again !!!! :huh:

The old server is still with the data, but the update experience was terrible :eek:hmy:

I recommend to try the updates in a test's machine and no in a production system!

Any body else knows something about this issue ?
Any solution ?

Thanks.
 

Bob

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#15

jgutierrez

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#16
hello unikbit,

I'm sorry for not replaying earlier, I was kinda busy...

The first time I logged into your server I saw that you had some extra features on "extensions", once you updated Elastix, did you updated again ferepbx or installed a new package or anything like that?

Now that I have logged in into your server I see that there isn't the menu for extensions, also in the Unembedded freepbx there are some errors and warnings...

I'll recommend you to make a fersh install of stable Elastix 1.3 branch
 

unikbit

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#17
Hello,

I did the fresh install, but still the inbound calls didn't work.

Afer I changed some details in Trunks at Peer Details is working now.

So in the Peer details default is:

host=sip.spunealo.com
username=myid
secret=mypass
type=peer

now I have changed

host=_sip.spunealo.com
username=_myid
secret=_mypass
type=peer

also User Details
secret=_mypass
type=_user
context=from-trunk

I don't know why but now is working.
If I put as default is not working

Thanks
Silviu
 

jgutierrez

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#18
mmm.. that is a little starnge....

From where did you took the idea of using "_" like a prefix for you register information? Did your provider expected that info like that???
 

unikbit

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#19
No, the provider don't expect this type of format.

I have thought to avoid
Peer details and user details to let only the registration line.

thanks,
 

jgutierrez

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#20
ok, thanks for your reply
 

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