SOLVED 3 problems, ? interrelated - trunks, calls and drop-outs (nodephone)...

[brainstorm]

Hi,
I have a couple of issues with my Incredible PBX – I don't know whether they are related, or separate issues. So for this I apologise in advance, and am happy to correct this query into separate forums if these issues are not related. I have searched the forums, and although there is threads about call drops, I have tried the suggestions, but to no avail.


I am running Incredible PBX 12.0.74, Asterisk Ver. 13.6.0 on a RaspberyPi 2.

The Trunk settings that follow were running faultlessly on my previous Incredible PBX box (asterisk version 11), and on a laptop – but upgrading it screwed up, and used this as an opportunity to experiment with the Raspberry Pi 2, as it automatically reboots after a power outage (few, but not uncommon here). Before anyone says it – it was cheaper to get the RPI2 than another battery for the lappy.

Trunk Name: Nodephone
PEER Details:
disallow=all
allow=alaw&ulaw&gsm
canreinvite=no
dtmfmode=rfc2833
fromdomain=sip.internode.on.net
username=XXXXXXXXXX
fromuser=XXXXXXXXXX
defaultuser=XXXXXXXXXX
host=sip.internode.on.net
secret=secretpass
trustrpid=no
sendrpid=no
qualify=yes
insecure=invite,port
canreinvite=no
type=peer
nat=yes
context=from-trunk

USER Context: XXXXXXXXXX
context=nodephone-outbound
host=sip.internode.on.net
secret=secretpass
type=user
username=XXXXXXXXXX

Register string:
XXXXXXXXXX:[email protected]/XXXXXXXXXX

This is where the first problem arises. Whenever the system reboots, I have to go into the system, change nothing but have to go into trunks, then submit changes and then apply config: then the trunk registers. It never drops out, always stays connected after I have done the aforementioned routine. It does not connect cleanly on reboot.

I have changed, in the peer details, the insecure=very to insecure=invite,port, as well as changing nat=no to nat=yes; same routine required. So, problem 1.

Problem 2:
I can make calls to any land line, be it local, interstate, special numbers (toll free/special toll etc), but I ring a mobile (cell) phone number, and although the mobile (cell) will ring, there is no ring tone that I can hear, when they answer there is no audio, and if I hang up from the pbx the mobile (cell) that I am ringing does not stop ringing – it does not disconnect. I have looked in the log files, which are below, but can't make out what is going on.

Log File:
[2016-01-09 22:40:36] WARNING[3417] res_odbc.c: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.1 Driver]MySQL server has gone away
[2016-01-09 22:40:36] WARNING[3417] res_odbc.c: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 5.1 Driver][mysqld-5.5.46-0+deb7u1]MySQL server has gone away (74)
[2016-01-09 22:40:36] WARNING[3417] res_odbc.c: SQL Execute error -1! Verifying connection to MySQL-asteriskcdrdb [MySQL-asteriskcdrdb]...
[2016-01-09 22:40:36] WARNING[3417] res_odbc.c: Connection is down attempting to reconnect...
[2016-01-09 22:40:36] WARNING[3414] func_cdr.c: CDR requires a value (CDR(variable)=value)
)[2016-01-09 22:40:41] NOTICE[3417] res_odbc.c: Connecting MySQL-asteriskcdrdb
[2016-01-09 22:40:41] NOTICE[3417] res_odbc.c: res_odbc: Connected to MySQL-asteriskcdrdb [MySQL-asteriskcdrdb]

and from the cdr-csv Log File:
"","SIP_num","044","from-internal",""""" <SIP_num>","SIP/150-00000040","SIP/Nodephone-00000041","Dial","SIP/Nodephone/044,300,Tt","2016-01-09 00:32:20",,"2016-01-09 00:32:23",2,0,"NO ANSWER","DOCUMENTATION","1452299540.436",""
"","SIP_num","mobile_num","from-internal",""""" <SIP_num>","SIP/156-00000042","SIP/Nodephone-00000043","Dial","SIP/Nodephone/mobile_num,300,Tt","2016-01-09 00:32:56",,"2016-01-09 00:33:17",20,0,"NO ANSWER","DOCUMENTATION","1452299576.446",""

Problem 3. Any land line call will disconnect when ringing out after 10 minutes. It is a clear call, which just terminates (log file for this attached). I have 6 extensions, and the behaviour is the same for all of them. The trunk does not drop. All I can find is that there is a hang-up, but where from?

Log File:
"2016-01-10 00:39:18","2016-01-10 00:39:18","2016-01-10 00:40:47",89,89,"ANSWERED","DOCUMENTATION","1452386358.546",""
"","SIP_num","dialled_num","from-internal",""""" <SIP_num>","SIP/156-0000004f","SIP/Nodephone-00000050","Dial","SIP/Nodephone/dialled_num,300,Tt","2016-01-10 10:20:06","2016-01-10 10:20:18","2016-01-10 10:30:57",651,639,"ANSWERED","DOCUMENTATION","1452421206.553",""
"","SIP_num","h","from-internal",""""" <dialled_num>","SIP/156-0000004f","","Hangup","","2016-01-10 10:30:57","2016-01-10 10:30:57","2016-01-10 10:30:57",0,0,"ANSWERED","DOCUMENTATION","1452421206.553",""

I had none of these issues on Asterisk 11, and had been successfully using it for over one year without any problems (loving it!!!). I was and am going through the same router and firewall (Fritz!box 7490).

Any input, or suggestions, would be greatly appreciated. I knew my way around ver11 quite well, but still trying to figure some of the finer details of ver 13, so please be patient if I respond asking for clarification. Also I am in Australia, so given the time difference I may take a while to respond.

Thanks

 

[ostridge]

I had none of these issues on Asterisk 11

Hi which GUI did you not have issues with asteriskGUI or IncredibleGUI ?

 

[brainstorm]

Hi which GUI did you not have issues with asteriskGUI or IncredibleGUI ?

IncredibleGUI.

 

[brainstorm]

Hi,
Thought I would post a small update.
I did a clean install of IncrediblePBX13-12.3raspbian, and only loaded the trunks section to see if this issue was resolved, and its behaviour is exactly the same; it requires me to hit submit changes and then apply config for it to register. Also interestingly I am getting 5 trunks registering, and yet only 1 trunk entry (as above). This is the same with both the old install described above and this clean install.

Thinking of trying on a laptop to see if I have the same issues....

 

[brainstorm]

OK then,
feel like I am flying alone on this one.

Thought I would give another update...
Installed IncrediblePBX13-12.2 on CentOS7 (minimal install).

Now I am getting the one of one trunk registering immediately. Hooray!!! Exactly the same trunk settings as the RPI2, running IncrediblePBX13-12.3raspbian. Go figure...

Now I have another problem, which I am trying to sort - I have to keep stopping and starting httpd service, and the gui keeps becoming unreachable. After much searching I found a suggestion from Ward stating to rename the index_custom.php file to index_custom_orig.php, which appears to have worked for some folks, but not on my system...jst have to keep doing service httpd stop then service httpd start. There appears no rhyme or reason behind the drops, but it sure is making getting this laptop up and running very painful...

Keep you in the loop with the outcome after I have played some more...

 

[brainstorm]

Maybe the last installment of this saga; far from resolved but I have a properly working PBX again.
So, what did I do (I ask to myself). As per the previous post, I installed IncrediblePBX13-12.2 on CentOS7 (minimal install), and I got trunks working correctly, but upon putting in extensions they wouldn't register, not matter what I attempted to do. So, having had the system down and no phones for a while - what do I do?

I have installed IncrediblePBX11-11.2centos - and guess what, using all of the aforementioned settings, I have a fully working PBX, now able to ring mobiles (cell phones), no dropouts, and trunks connecting immediately. So what has changed between all of these versions, or more to the point, what are the differences between 11 and 13 to have such a drastic impact on performance?

 

[brainstorm]

OK, I spoke too soon. Everything works, except there are still the dropouts after 10 minutes - log files just show the call is terminated...

I have checked all the NAT settings; have added a line session-timers=refuse in the SIP settings; and now have adjusted so that I am using reinvite both in the SIP settings and on the extensions. I will keep posting until this gets resolved, there may be some useful information that appears for someone else.

I have also previously wound out the session time in the SIP settings, but I have removed this as it did nothing.

If anyone else has some pearls, please feel free to chime in...

 

[brainstorm]

After much playing, I think this issue is now resolved - Just one more lot of testing, and then I will be happy to lay this issue to bed.

Firstly, I would like to thank Ryan at Internode, who took the time to explore and decipher the SIP logs at their side, to determine if it was the PBX that was terminating the calls, or the internode SIP server - from the logs it was found that the internode SIP server was terminating the call because there was no acknowledgement from the PBX to keep the line alive - that is my PBX is not responding to their server to keep the call alive.

I first took a look at my modem/router/firewall - a Fritz!box 7490. I gave this thought as it has SIP capabilities onboard, and I had read somewhere that port 5060 was considered a port which is held in reserve by the F!B because of its SIP capabilities. So I went to a different modem/router first to rule this issue out. Unfortunately (or fortunately) this proved fruitless as the calls still terminated after 10:22.

Then I went searching in these forums and Google - I coudn't find a hint anywhere - then I came across a link in these forums, and Ward Mundy had suggested this to another issue titled AnveoDirect Calls Drops 15m32s back in 2013:

"Add to the trunk settings (first one, then first two, then all three) to the trunk settings
canreinvite=yes
dtmfmode=info
qualify=yes"

I left my dtmfmode=rfc2833, but confirmed the other two settings, which had changed and varied through the time of problem solving this, and now low and behold, calls longer than the 10:22 - hooray.

I will do a final post after the final testing tonight - which also involves confirming a response to the SIP server at internode.

 

[brainstorm]

OK then,
this all works really well. I have incoming and outgoing calls (wasn't getting incoming calls as well, wondered why it was so quiet here for the last few days) - outgoing required the insecure=invite. Calls are now no longer only 10.22 long. So, for completion, and just as a reference for anyone else that is having issues (very little reference of the trunk settings for internode - and these needed the little tweaks to get it working). If not internode, I got these tips after reading many examples of trunks - for the gurus this might all be obvious - and although not a noob, I did trip up on this a little (as I was using settings that were working - don't know why things changed...) Anyhow...

Nodephone:
disallow=all
allow=alaw
username=08XXXXXXXX
defaultuser=08XXXXXXXX
fromuser=08XXXXXXXX
secret=my_password
type=peer
qualify=yes
insecure=port.invite
host=sip.internode.on.net
fromdomain=sip.internode.on.net
dtmfmode=rfc2833
canreinvite=yes
context=from-trunk
port=5060

08XXXXXXXX:
type=peer
username=08XXXXXXXX
secret=my_password
host=sip.internode.on.net
context=from-trunk
insecure=invite

Register String
08XXXXXXXX:my_password@sip.internode.on.net

So, hopefully this might help someone...but my initial questions remained unanswered. Why could I not ring mobile (cell) phones, and why with asterisk13 (for RPi) did I have to apply changes to the trunk settings after a reboot to allow incoming and outgoing calls? I am intrigued to know this - although now I have a working PBX. What a long journey, and at least this forum helped me get my thoughts out there...

Now to play with incredible PBX 11 for RPi, see if I can get this to work (but not straight away... I think I need a break...)