[SOLVED] Misc Destination Suddenly Ending Calls

When I use Putty and input the above referenced command “grep” the system stalls then kicks me out of putty… I believe that I made that clear earlier…

I think you misunderstood what I was saying about bashing platforms… I am a Microsoft person when you refer to Windows as “windoze” I find that offensive. No matter how frustrated I get with Freepbx, Asterisk , or Linux I don’t bad mouth them nor do I the people working on the problems behind them, I would appreciate the same from you.

Is this thread loosing focus and productivity. It may be time to lock this

About badmouthing, That would be a little hard for me to change my attitude about, I am pretty damn sure I have been using Microsoft products for a lot longer than you (how old are you?) , it is not a religious thing with me, just one of learned contempt, frustration and wasted money but that is what you have chosen so YMMV. (but I am sure many here will agree with me :slight_smile: )

About grep, you are just plain doing it wrong. there is no way a user space program (grep) in linux can kill the underlying calling process (bash, ssh, scp or anything else, (putty and winscp are just wrappers around ssh) unless your kernel is seriously damaged. It just doesn’t happen in a real OS like linux.

grep -E “123457|123456” /var/log/asterisk/full

will invariably “grep out” any line with 123457 or 123456 anywhere in it.

grep -i error /var/log/asterisk/full

will similarly extract any line with error in it (case ignored), all you need to do is

man grep

for the syntax. (or for that matter “man bash” or “man scp” or pretty much anything else in linux, man is short for manual)

I understand your frustration as you are coming from a Micro$oft background and we just “don’t do it your way here”, but if so why not just use Lync and save yourself a whole bunch of time, (maybe not money though :slight_smile: )

It took me almost an hour to extract some log info… When I run putty, I can log in but as soon as I enter a command the system stops in its tracks and then kicks me out. I did this friday when people were in the office and it garbled their calls…

If I could get my company to use Lynx or 3cx I would but I had to Pay out of my own pocket for Freepbx support time…

https://drive.google.com/file/d/0B9Oae_c86t9fd1VsNExLZ1ozdDg/edit?usp=sharing

Why would you think Lynx or 3CX is a better product than Asterisk and FreePBX?

You are having very unusual issues. SSH to a system should not effect call quality. Makes me wonder if you hardware or network is stable.

Let’s look at the errors in your log.

Your custom WAV file was recorded at the wrong sampling rate? Did you see the notice in the recordings module that the file must be 8Khz mono?

layback(“SIP/Nextiva-000001fe”, “custom/Test,noanswer”) in new stack
[2014-07-06 23:18:16] WARNING[19709][C-000000ce] format_wav.c: Unexpected frequency mismatch 44100 (expecting 8000)

You also do things the hard way. When I said a short blank file I meant to simply use the recordings module to make one from an extension not to upload something you created on your computer. Since Asterisk could not play the file I assume the channel wasn’t answered so the test criteria was not met.

Also did you make any changes to config files? This is not a NANP address 6788005798418 and it looks like you have a loop in the routine. The only reference I can find to sub-flp is in the custom context code. Are you trying to use a custom context and if so why?

It goes though that look and actually looks like Asterisk crashed.

It is also silly that you had to pay for support out of your own pocket. That’s just my opinion though.

You need to tell us about any customization you have done to this system.

You really need to start over, your nextiva trunk and your outbound routes are reantrant, you are trying to dial

4708005798418

but as you see that is not working. , I presume you are trying to dial

8005798418

so look at your endpoint 470 and your outbound routes and correct as necessary.

but this clip gives a lot away

[2014-07-06 23:18:16] WARNING[19709][C-000000ce] format_wav.c: Unexpected frequency mismatch 44100 (expecting 8000)
[2014-07-06 23:18:16] WARNING[19709][C-000000ce] file.c: Unable to open format wav
[2014-07-06 23:18:16] WARNING[19709][C-000000ce] file.c: Unable to open custom/Test (format (ulaw)): No such file or directory
[2014-07-06 23:18:16] WARNING[19709][C-000000ce] app_playback.c: ast_streamfile failed on SIP/Nextiva-000001fe for custom/Test,noanswer
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@app-announcement-1:6] Goto(“SIP/Nextiva-000001fe”, “ext-miscdests,2,1”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Goto (ext-miscdests,2,1)
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [2@ext-miscdests:1] NoOp(“SIP/Nextiva-000001fe”, “MiscDest: Service”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [2@ext-miscdests:2] Goto(“SIP/Nextiva-000001fe”, “from-internal,8005798418,1”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Goto (from-internal,8005798418,1)
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [8005798418@from-internal:1] Macro(“SIP/Nextiva-000001fe”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:1] Set(“SIP/Nextiva-000001fe”, “TOUCH_MONITOR=1404703084.510”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:2] Set(“SIP/Nextiva-000001fe”, “AMPUSER=7706864372”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:3] GotoIf(“SIP/Nextiva-000001fe”, “0?report”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:4] ExecIf(“SIP/Nextiva-000001fe”, “0?Set(REALCALLERIDNUM=7706864372)”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:5] Set(“SIP/Nextiva-000001fe”, “AMPUSER=”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:6] GotoIf(“SIP/Nextiva-000001fe”, “0?limit”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:7] Set(“SIP/Nextiva-000001fe”, “AMPUSERCIDNAME=”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:8] GotoIf(“SIP/Nextiva-000001fe”, “1?report”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Goto (macro-user-callerid,s,16)
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:16] GotoIf(“SIP/Nextiva-000001fe”, “1?continue”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Goto (macro-user-callerid,s,30)
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:30] Set(“SIP/Nextiva-000001fe”, “CALLERID(number)=7706864372”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:31] Set(“SIP/Nextiva-000001fe”, “CALLERID(name)=Cell Phone GA”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:32] Set(“SIP/Nextiva-000001fe”, “CDR(cnum)=7706864372”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:33] Set(“SIP/Nextiva-000001fe”, “CDR(cnam)=Cell Phone GA”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [s@macro-user-callerid:34] Set(“SIP/Nextiva-000001fe”, “CHANNEL(language)=en”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [8005798418@from-internal:2] ExecIf(“SIP/Nextiva-000001fe”, “0 ?Set(CDR(accountcode)=)”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [8005798418@from-internal:3] Set(“SIP/Nextiva-000001fe”, “MOHCLASS=default”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [8005798418@from-internal:4] ExecIf(“SIP/Nextiva-000001fe”, “0?Set(TRUNKCIDOVERRIDE=587544469)”) in new stack
[2014-07-06 23:18:16] VERBOSE[19709][C-000000ce] pbx.c: – Executing [8005798418@from-internal:5] Set(“SIP/Nextiva-000001fe”, “_NODEST=”) in new stack

you have custom destinations, this and that, you are trying to use audio files that are in the wrong format , the list goes on.

There is a wiki here for newbies, you should start there until you get the basics working.

As to why putty isn’t working, that is NOT a FreePBX problem.

OK to start from scratch.

I downloaded the distro pack, burned it to a CD installed it to a i3 machine with 4bg ddr3. Launched freepbx and purchased endpoint manager. Configured the sips as was instructed by nextiva, and set the phones in endpoint manager. Read a bit in the wiki for misc destination and bam system is up and running fine with no problems for a few months. The only thing that’s been having issues is a single polycom ip650 starts going haywire around 4:55pm every day (it starts to reset but never finishes, simply power it off and back on and its fine again) and the second issue is that on random internal calls (ext to ext) there is no audio (happens less than once a day)

As to why this system is doing what it’s doing is beyond me.

As for the sound file, I can’t make one from a phone as I’m at home, used the computer and made one then realized the quality issue after the fact. Made a new one, uploaded it and haven’t been able to try it yet.

I’m pretty sure your brief reading of misc destination was inadequate , few would need that as a destination, but please post what it is.

( a convenient short file of silence would possibly be /var/lib/asterisk/sounds/en/silence/1.wav, just cp it where you need it)

http://wiki.freepbx.org/display/F2/Misc+Destinations

I opened the module, put the 800 number in one box and a name in the other and saved it. Went over to the the ring group and clicked misc dest and the name at the bottom and saved it. worked fine…

I disagree as to it working fine, you just tried calling a variant of it a thousand times :slight_smile:

It worked fine until last Wednesday.

OK, So I deleted the trunk, and the OB route and re added them and that has fixed the problem…

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I am glad we got through this and no one lost an eye :wink:

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