Vicidial reports "Dial Timed Out" on answered call

[jsmith]

I'll admit upfront I'm a Vicidial newbie (but a long-time Asterisk user/developer/contributor). I've setup Vicidial for the first time, and it seems to be mostly working, except that when Vicidial dials a lead and the other person answers the call, Vicidial doesn't seem to notice, and after 50 seconds or so Vicidial reports something like "Dial timed out. Contact your system administrator". (Sorry, I don't have the exact text in front of me.)

Can anyone help me figure out what's going on?

I'm running Vicidial 2.0.2b3 (will upgrade to 2.0.2 tonight). Everything else seems to be working (except for DTMF from within a MeetMe, but that's another fight for another day.)

[mflorell]

Please post results of "screen -r"

Do you have the "o" flag in your Dial command?


As for DTMF in meetme, it is only possible using all Zaptel channels.

[jsmith]

Thanks for your quick response!

Please post results of "screen -r"

[root@mushroom ~]# screen -r
There are several suitable screens on:
7112.ASTfastlog (Detached)
27953.ASTsend (Detached)
1541.pts-1.mushroom (Attached)
27962.ASTVDremote (Detached)
27950.ASTupdate (Detached)
27956.ASTlisten (Detached)
7109.ASTVDadapt (Detached)
27959.ASTVDauto (Detached)
27804.asterisk (Detached)

Do you have the "o" flag in your Dial command?

Nope... I don't. Is this really necessary? We're using this dialplan for other dialing besides what ViciDial uses, so I'm not sure I want the "o" option on all my outbound dialing. (It's my understanding that the "o" flag is to bring back Asterisk's old/broken Caller*ID functionality when a call is transferred.) Mind explaining why this is necessary?

As for DTMF in meetme, it is only possible using all Zaptel channels.

That's too bad... our agents are using softphones, and they need to be able to navigate through IVR systems to reach the correct respondant. Any ideas on how this might be accomplished?

-Jared

[mflorell]

The "o" flag is required for VICIDIAL usage. And you don't have to use it for all outbound calls, simply add another outbound Dialing set of extens in your dialplan and add a prefix number to it, then use the campaign dial prefix field to have the campaign dial out over those extens.

As for DTMF, you can always use DTMF macros and campaign DTMF presets.

[jsmith]

The "o" flag is required for VICIDIAL usage.

Got it. That fixed our "dial timed out" problem. Thank you so much!

As for DTMF, you can always use DTMF macros and campaign DTMF presets.

I've looked through the campaign DTMF presets, but nothing jumped out at me. Mind elaborating on what you mean by "DTMF macros"?

-Jared

[aster1]

there is a send dtmf box in vicidial agent interface u can type number in their and click send dtmf to browse through ivr ..

[aster1]

there is a send dtmf box in vicidial agent interface u can type number in their and click send dtmf to browse through ivr ..

[jsmith]

there is a send dtmf box in vicidial agent interface u can type number in their and click send dtmf to browse through ivr ..

Yes, I've tried that, and it doesn't work either. I'd be happy to use that, if I could get it to work.. Any ideas on why that isn't working either?

-Jared

[mflorell]

Can you post Asterisk CLI output from when you try to use SendDTMF?

[jsmith]

Can you post Asterisk CLI output from when you try to use SendDTMF?

I'll capture that later today, and post it here.

-Jared

[jsmith]

mushroom*CLI> ^M -- Executing Answer("Local/8500998@default-028e,2", "") in new stack
mushroom*CLI> ^M -- Executing Playback("Local/8500998@default-028e,2", "silence") in new stack
> Channel Local/8500998@default-028e,1 was answered.
mushroom*CLI> ^M -- Executing MeetMe("Local/8500998@default-028e,1", "8600051|q") in new stack
mushroom*CLI> ^M -- Playing 'silence' (language 'en')
mushroom*CLI> ^MFeb 28 14:42:01 WARNING[8250]: file.c:1041 ast_waitstream: ^@Unexpected control subclass '-1'
mushroom*CLI> ^M -- Executing AGI("Local/8500998@default-028e,2", "agi-dtmf.agi") in new stack
mushroom*CLI> ^M -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-dtmf.agi

-Jared

[mflorell]

What version of Asterisk are you using?

I have not seen that error in a SendDTMF before.

[jsmith]

What version of Asterisk are you using?

I'm using the latest version of Asterisk 1.2 from the 1.2 branch in SVN.

-Jared

[aster1]

Try with asterisk 1.2.14 its working for me with that version .

[jsmith]

Try with asterisk 1.2.14 its working for me with that version .

Unfortunately, it's not that simple... Due to some other problems with older versions of Asterisk that have since been fixed, I'm forced to use the latest Asterisk 1.2 from SVN.

-Jared

[mflorell]

Which specific errors in 1.2.14 are fixed in current SVN? Which SVN build?

[jsmith]

Which specific errors in 1.2.14 are fixed in current SVN? Which SVN build?

Lots of problems (I don't remember all of them), but by now I'm sure you know that everyone should upgrade to 1.2.16 thanks to a serious problem in the SIP channel.

Now, in going through my dialplan I've noticed that agi-dtmf is getting called through the regular AGI mechanism, and not through FastAGI. Should I change that? I can't find instructions anywhere on how to call agi-dtmf.agi through FastAGI. Would that cause the problems I'm seeing?

[mflorell]

there are several AGIs that are still called through the AGI mechanism. They are also not called anywhere near as often as the call_log and VDhangup scripts were. Eventually more scripts will be converted to FastAGI, but the performance gains will be nowhere near what they were moving the call_log scripts over.

[jsmith]

there are several AGIs that are still called through the AGI mechanism.

Interesting... in doing more debugging this morning, agi-dtmf.agi seems to be hanging before completing... I'll try to get an "agi debug" trace.

[jsmith]

OK, I have an AGI debug trace now:

-- Executing Answer("Local/8500998@default-51d7,2", "") in new stack
-- Executing Playback("Local/8500998@default-51d7,2", "silence") in new stack
-- Executing MeetMe("Local/8500998@default-51d7,1", "8600051|q") in new stack
-- Playing 'silence' (language 'en')
Mar 8 12:00:17 WARNING[16571]: file.c:1041 ast_waitstream: Unexpected control subclass '-1'
-- Executing AGI("Local/8500998@default-51d7,2", "agi-dtmf.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-dtmf.agi
AGI Tx >> agi_request: agi-dtmf.agi
AGI Tx >> agi_channel: Local/8500998@default-51d7,2
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Local
AGI Tx >> agi_uniqueid: 1173380417.4271
AGI Tx >> agi_callerid: 5
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: default
AGI Tx >> agi_extension: 8500998
AGI Tx >> agi_priority: 3
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Environment Dump:
-- accountcode =
-- callerid = 5
-- calleridname = unknown
-- callingani2 = 0
-- callingpres = 0
-- callingtns = 0
-- callington = 0
-- channel = Local/8500998@default-51d7,2
-- context = default
-- dnid = unknown
-- enhanced = 0.0
-- extension = 8500998
-- language = en
-- priority = 3
-- rdnis = unknown
-- request = agi-dtmf.agi
-- type = Local
-- uniqueid = 1173380417.4271


At that point, the AGI script just hands, and doesn't continue. Any ideas?

[jsmith]

While I'm on the topic... is it just me or is there nothing at all to do with DTMF in agi-dtmf.agi?

If I'm reading it correctly, it simply streams the digits from the callerid into the channel. But not the DTMF digits... the voice of Allison saying "one", "two", etc.

Am I going crazy? Am I missing something fundamental here? Can anyone enlighten me?

[mflorell]

Are the sounds files from astguiclient in the proper places on your system?

Included here are the DTMF audio files that the dtmf script uses to play the dtmf tones.

What is the result when you do "show version" in Asterisk CLI?

[jsmith]

I was finally able to solve my DTMF problems by rewriting the agi-dtmf.agi script. I hope I don't offend anyone when I say it was poorly written. In short, the stock AGI script was waiting for a response from Asterisk when it shouldn't have been, and wasn't waiting for a response from Asterisk when it should have been. It now works as it should. (I'd be happy to post my changes somewhere if anyone is interested.)

-Jared

[enjay]

post post it on tracker..

[jsmith]

post post it on tracker..

Posted as tracker issue #88

[mflorell]

No offense taken

Can you tell me a little more about the problems you had with the dtmf AGI script?

All I can see in your posted code is several TEST VERBOSE prints and some $result = <STDIN>; added after commands.

I haven't really done much to that script in quite a while, some of the code in some of these AGI scripts handling callerID is left in or duplicated so that it is backwards-compatible with the old 1.0 tree of Asterisk.

I have never had any issues with the dtmf AGI script and I was wondering exactly what was happening on your system.

What Linux Distro?
What Perl version?
What asterisk-perl module version?
What Asterisk version?

[jsmith]

Can you tell me a little more about the problems you had with the dtmf AGI script?

All I can see in your posted code is several TEST VERBOSE prints and some $result = <STDIN>; added after commands.

The old code wrote a bunch of things to STDERR, but this is only visible from the *first* console that's started, so you'll never see this debug info. I replaced it with the VERBOSE command, so that the debug info will show up in all of the Asterisk consoles. (You set the verobsity level to 1... so the debug will show up if you have your CLI verbosity level at 1 or higher. You may want to change that to 3 or 4 or even higher.)

The main problem I was having was that the AGI script would hang... the script was expecting Asterisk to send something, and Asterisk was waiting for the script to send something. It appeared to be buried somewhere in the Asterisk::AGI module (as it happened when you tried to use the GET VARIABLE command, but it's before that command was sent to Asterisk). I simply ripped out all the Asterisk::AGI commands and replaced them with raw calls.

While I was investigating that problem, I noticed that the script sent lots of commands to Asterisk without reading the response (especially during the STREAM FILE commands), so I made sure we read back the response after each of those commands as well.

What Linux Distro?
What Perl version?
What asterisk-perl module version?
What Asterisk version?

[root@mushroom ~]# cat /etc/redhat-release
CentOS release 4.4 (Final)

[root@mushroom ~]# uname -a
Linux mushroom 2.6.9-42.0.3.ELsmp #1 SMP Fri Oct 6 06:28:26 CDT 2006 x86_64 x86_64 x86_64 GNU/Linux

[root@mushroom ~]# perl -v
This is perl, v5.8.5 built for x86_64-linux-thread-multi

asterisk-perl-0.09

[root@mushroom ~]# asterisk -rx 'show version'
Asterisk SVN-branch-1.2-r55434M built by root @ mushroom on a x86_64

[mflorell]

asterisk-perl 0.09 is the issue, there are other issues with 0.09 that were not issues with 0.08 like the call_status response.

[enjay]

Matt,

Do you not recommend using asterisk-perl 0.9?

[mflorell]

I do not recommend using the asterisk-perl-0.09 module. I mention this several times in various documents. There are two issues that I know of, not to mention the added complexity that is unneeded that is written into 0.09. The 0.08 version has been tested on every version of Asterisk going back 3 years and is fast and solid with VICIDIAL.

[ckwall]

I am having the same issue. However I am using asterisk-perl-0.08. is there another possible work around? I am not the genius that jsmith is and would not know where to begin to rewrite the code...

[ckwall]

I have downloaded and replaced the agi-dtmf.agi from the bug posted by jsmith. however I am still having issues. Now there are no tones being sent. Before I replaced the file, it would give me just a short blip of a tone. Now after replacing the file, I get nothing.

[devafree]

I am giving a very simplistic reply but one that solved the trouble of very short blip for me. I prefixed and suffixed a comma to each digit . (This was on a old install, now with 2.0.2b3 on asterisk 1.2.14, i have no troubles (except i cannot send lengthy DTMF on SIP with g729a codec , as per my earlier post - thats not resolved as i am forced by the ISP to use only SIP and g729).

I was wondering is jsmith is the co-author of *- TFOT ?

yep thats all i am contributing.

regards
devafree

[jsmith]

I was wondering is jsmith is the co-author of *- TFOT ?

Guilty as charged Speaking of which, we're working on the second edition of TFOT. I have a couple of minor Vicidial issues to get resolved, and then I'll have more time to work on it.

-Jared

[devafree]

Very honoured to meet you. Your book got me started on Asterisk , I couldn't figure out head or tail then ( very little better now i suppose , heh). Thats probably how i am here (coz gnudialerdidn't give a scratch install doc, )

Thank you very much.

devafree

[ckwall]

The comma thing did not work.
I am on asterisk version 1.2.15
vicidialer version (from the admin section) VERSION: 2.0.85 BUILD: 70206-1323
Linux version RedHat Enterprise 4.4