Hi,
CentOS 5 2.6.18-194.26.1.el5 x86_64
Asterisk 1.8.0
FreePBX 2.8.0.4
Directory 2.8.0.0
I have setup Directory with 7 extensions. When I call and enter a matching name, the system shuts up and never puts the call through:
– directory.agi,dir=2: Found 1 possible matches from 727
– directory.agi,dir=2: Found the following matches:
– directory.agi,dir=2: name: Screen Phone, audio: vm, dial: 5101
Any hints please?
Cheers guys
http://help.freepbx.org/forum/freepbx/users/flite-says-display-name-of-user-followed-by-the-word-bar#comment-36559 gives a fix…
However, Directory still dumps a call sometimes. Like if Announce Extension is turned on. Will test that more when I have time…
Tried re-opening the bug report, but I guess I can’t…
http://www.freepbx.org/trac/ticket/4750#comment:21
There is also still the issue of text to speech appending bar hash to the end of each spoken name.
Cheers
Dave
Thanks guys. That gets my calls put through!
Your welcome 
Thank you all. I updated from my patched 2.8.0.1 to 2.8.0.3 and it is working.
trying to load the 2.9 version of them module onto a 2.8 base will probably cause problems as you noticed.
If the upgrade to 2.8.0.2 doesn’t fix your bug, please file a trac ticket per mbrevda’s request above with exact steps to reproduce the issue (what the entry should look like, what sequence of steps the caller should take to create the problem.)
thanks.
Guys,
There definitely was a bug with the way directory listened while waiting for dtmf after reading the list of matches. This issue has been fixed in 2.9 (and I believe back ported to 2.8). Please update your modules (or test the alpha taball if you have a development/test system) and open a bug with exact steps to reproduce IF YOU STILL SEE AN ISSUE.
OK Philippe,
I hope I’ve posted enough info over in ticket 4786
Thanks for your help guys!
Hi,
I updated to FreePBX 2.8.1.0 & Directory 2.8.0.2 (the latest available in Module Admin) and the problem is just the same.
I tried to upgrade to 2.9.0.0 by Module Upload from the trac. Oddly it showed the message “Directory 2.8.0.2 will be upgraded to 2.8.0.2”. Once installed I had a Migrated Directory as well as my test directory. But all links on that page led to a blank html page!
Tried 2.9.0.1. It showed the message “Directory 2.9.0.0 will be upgraded to 2.8.0.2”! That orange upgrade screen got as far as “Please wait while module actions are performed” and hung. Module Admin shows “Disabled; Pending upgrade to 2.9.0.1”
Downgraded to 2.8.0.2 and I’m back where I started with the original issue (steps to create earlier in this thread).
Anyone had success with curing this? (Tadpole question: Do I need to take the 2.9 FreePBX to use this? Doh!)
Hey. I had the same problem and couldn’t find the answer so I decided to dig a little in the code.
I found that line 185 in directory.lib.php would block forever if $keys is empty ($keys==’’).
So I changed it from:
$ret=$this->agi->evaluate('SAY ALPHA ‘.$char.’ '.$keys);
to:
$ret=$this->agi->evaluate('SAY ALPHA ‘.$char.’ '.$keys."|");
to make sure the 2nd parameter to SAY ALPHA isn’t empty.
This is probably not the proper fix, but at least it is working now. From there, one developer will maybe help us.
Just wondered if you or someone else could please help me with this…
Happy Christmas all!
If I set the Name Announcement to Text to Speech, the call goes through (with text to speech announcement (with the word bar | on the end of the name)).
If I set the Name Announcement to Voicemail Greeting the call goes through (with the spoken name) if the greeting exists. However it the greeting hasn’t been recorded, the call is not put through (silence until hangup)!
If I set the Name Announcement to Spell Name it fails (silence until hangup). This has worked in the past and nicely chosen Voicemail Greeting if one was available or falls back to Spell if not available. However, since I now have speech working the nice fallback system seems to be broken. I would assume it would prefer Voicemail, then Speech, then Spell?
Is this in directory.agi? I’m guessing something is going wrong in the //playback entries section.
Thanks for any help!
Thanks for reply. I have been skiing for a week, so sorry for delay. I’m not sure this shows any more details. Please tolerate me if you asked for something else. I appreciate your time.
core set verbose 5
Verbosity was 3 and is now 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [44xxxxxxxxxx@from-sip-external:1] NoOp(“SIP/194.xxx.xxx.xxx-00000087”, “Received incoming SIP connection from unknown peer to 44xxxxxxxxxx”) in new stack
– Executing [44xxxxxxxxxx@from-sip-external:2] Set(“SIP/194.xxx.xxx.xxx-00000087”, “DID=44xxxxxxxxxx”) in new stack
– Executing [44xxxxxxxxxx@from-sip-external:3] Goto(“SIP/194.xxx.xxx.xxx-00000087”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/194.xxx.xxx.xxx-00000087”, “1?checklang:noanonymous”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] GotoIf(“SIP/194.xxx.xxx.xxx-00000087”, “0?setlanguage:from-trunk,44xxxxxxxxxx,1”) in new stack
– Goto (from-trunk,44xxxxxxxxxx,1)
– Executing [44xxxxxxxxxx@from-trunk:1] Set(“SIP/194.xxx.xxx.xxx-00000087”, “__FROM_DID=44xxxxxxxxxx”) in new stack
– Executing [44xxxxxxxxxx@from-trunk:2] ExecIf(“SIP/194.xxx.xxx.xxx-00000087”, “1 ?Set(CALLERID(name)=08453677500)”) in new stack
– Executing [44xxxxxxxxxx@from-trunk:3] Set(“SIP/194.xxx.xxx.xxx-00000087”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [44xxxxxxxxxx@from-trunk:4] Set(“SIP/194.xxx.xxx.xxx-00000087”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [44xxxxxxxxxx@from-trunk:5] Goto(“SIP/194.xxx.xxx.xxx-00000087”, “ivr-2,s,1”) in new stack
– Goto (ivr-2,s,1)
– Executing [s@ivr-2:1] Set(“SIP/194.xxx.xxx.xxx-00000087”, “MSG=custom/Reception_Intro”) in new stack
– Executing [s@ivr-2:2] Set(“SIP/194.xxx.xxx.xxx-00000087”, “LOOPCOUNT=0”) in new stack
– Executing [s@ivr-2:3] Set(“SIP/194.xxx.xxx.xxx-00000087”, “__DIR-CONTEXT=”) in new stack
– Executing [s@ivr-2:4] Set(“SIP/194.xxx.xxx.xxx-00000087”, “_IVR_CONTEXT_ivr-2=”) in new stack
– Executing [s@ivr-2:5] Set(“SIP/194.xxx.xxx.xxx-00000087”, “_IVR_CONTEXT=ivr-2”) in new stack
– Executing [s@ivr-2:6] GotoIf(“SIP/194.xxx.xxx.xxx-00000087”, “0?begin”) in new stack
– Executing [s@ivr-2:7] Answer(“SIP/194.xxx.xxx.xxx-00000087”, “”) in new stack
– Executing [s@ivr-2:8] Wait(“SIP/194.xxx.xxx.xxx-00000087”, “1”) in new stack
– Executing [s@ivr-2:9] Set(“SIP/194.xxx.xxx.xxx-00000087”, “TIMEOUT(digit)=3”) in new stack
– Digit timeout set to 3.000
– Executing [s@ivr-2:10] Set(“SIP/194.xxx.xxx.xxx-00000087”, “TIMEOUT(response)=10”) in new stack
– Response timeout set to 10.000
– Executing [s@ivr-2:11] Set(“SIP/194.xxx.xxx.xxx-00000087”, “__IVR_RETVM=”) in new stack
– Executing [s@ivr-2:12] ExecIf(“SIP/194.xxx.xxx.xxx-00000087”, “1?Background(custom/Reception_Intro)”) in new stack
– <SIP/194.xxx.xxx.xxx-00000087> Playing ‘custom/Reception_Intro.slin’ (language ‘en’)
– Executing [1@ivr-2:1] NoOp(“SIP/194.xxx.xxx.xxx-00000087”, "Deleting: ") in new stack
– Executing [1@ivr-2:2] Set(“SIP/194.xxx.xxx.xxx-00000087”, “__NODEST=”) in new stack
– Executing [1@ivr-2:3] Goto(“SIP/194.xxx.xxx.xxx-00000087”, “directory,2,1”) in new stack
– Goto (directory,2,1)
– Executing [2@directory:1] Answer(“SIP/194.xxx.xxx.xxx-00000087”, “”) in new stack
– Executing [2@directory:2] Wait(“SIP/194.xxx.xxx.xxx-00000087”, “1”) in new stack
– Executing [2@directory:3] AGI(“SIP/194.xxx.xxx.xxx-00000087”, “directory.agi,dir=2”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/directory.agi
– <SIP/194.xxx.xxx.xxx-00000087> Playing ‘cdir-welcome.slin’ (language ‘en’)
– <SIP/194.xxx.xxx.xxx-00000087> Playing ‘cdir-please-enter-first-three.slin’ (language ‘en’)
– directory.agi,dir=2: Found 1 possible matches from 328
– directory.agi,dir=2: Found 1 possible matches from 328
– directory.agi,dir=2: Found the following matches:
– directory.agi,dir=2: name: Dave Xxxxxx, audio: vm, dial: 5103
– <SIP/194.xxx.xxx.xxx-00000087>AGI Script directory.agi completed, returning -1
– Executing [h@directory:1] Macro(“SIP/194.xxx.xxx.xxx-00000087”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/194.xxx.xxx.xxx-00000087”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/194.xxx.xxx.xxx-00000087”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/194.xxx.xxx.xxx-00000087”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] Hangup(“SIP/194.xxx.xxx.xxx-00000087”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/194.xxx.xxx.xxx-00000087’ in macro ‘hangupcall’
== Spawn extension (directory, h, 1) exited non-zero on ‘SIP/194.xxx.xxx.xxx-00000087’
I held listening to silence for 20 seconds or so before hanging up. The destination extension 5103 never rang.
Cheers
Dave
Your going to have to show us a little bit more than that. Can we get a complete call trace please?
Hello,
I’ve raised a ticket describing a variant on the hangup issue as promised (over two months ago)! 
I thought development work could be slow - seems end users can move slower yet…
Thanks
Dave