that is an invalid conference number

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that is an invalid conference number

Postby rdelrosario » Wed Jul 25, 2007 5:05 pm

I went throught the VICI scratch intsall and with the exception of installing ZAP related sections of the instructions, I followed everything "I THINK" to a tee.

When I login to a campaign. My agent softphone rings and when I answer it, it states that "that is an invalid conference number". I looked in the vici conference numbers in the gui and it seems to be there. It was currently using the 8600051 I think.

two things.. Can someone explain to me how/why the conference numbers are used. Am I correct in stating that when you login to vici as an agent that asterisk call that agent up and when they answer it puts them in a conference room to then send them phone numbers to dial whether you manually or auto dial?

I thought at first my problem was that I am not using zaptel drivers and didn't install the zt-dummy stuff. I'm using pure IAX trunking with voicepulse. Everything works manually when I dial from my softphone through asterisk and internal phone ext. to ext works fine. Its ony through vici that it doesn't work.

Any information to where I can find out why this problem is happening would be greatly appreciated. I'm very very new to this so be kind... Actually I have a 1 weeks experience with Asterisk and Linux.

ThankS!!
rdelrosario
 
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here is the asterisk console messages when the above happens

Postby rdelrosario » Wed Jul 25, 2007 5:27 pm

-- Registered SIP '5981' at 192.168.50.125 port 23146 expires 3600
-- Saved useragent "X-Lite release 1011s stamp 41150" for peer 5981
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/5981-081b0310 was answered.
-- Executing MeetMe("SIP/5981-081b0310", "8600069") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Manager 'sendcron' logged off from 127.0.0.1
Jul 25 18:23:39 WARNING[417]: chan_zap.c:1064 zt_open: Unable to open '/dev/zap/pseudo': No such device
Jul 25 18:23:39 ERROR[417]: chan_zap.c:7551 chandup: Unable to dup channel: No such device
Jul 25 18:23:39 WARNING[417]: app_meetme.c:464 build_conf: Unable to open pseudo channel - trying device
Jul 25 18:23:39 WARNING[417]: app_meetme.c:467 build_conf: Unable to open pseudo device
-- Playing 'conf-invalid' (language 'en')
== Spawn extension (outgoing, 8600069, 1) exited non-zero on 'SIP/5981-081b0310'
-- Executing DeadAGI("SIP/5981-081b0310", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("SIP/5981-081b0310", "agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16---------------)") in

new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0
phone*CLI>




I notice that is states zap warnings and specifically app_meetme stuff which I don't recall enabling... does this point out where my problem is?

Please help
rdelrosario
 
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more info about my error above

Postby rdelrosario » Wed Jul 25, 2007 6:13 pm

here is another output from asterisk console from a different extension number. a little bit different error message


> Channel SIP/5933-081aaad0 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
== Starting SIP/5933-081aaad0 at default,8600069,1 failed so falling back to exten 's'
== Starting SIP/5933-081aaad0 at default,s,1 still failed so falling back to context 'default'
Jul 25 18:21:37 WARNING[32695]: pbx.c:2377 __ast_pbx_run: Channel 'SIP/5933-081aaad0' sent into invalid extension 's' in context 'default', but no invalid handler
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1


The one above this (post prior to this) was for an extension that actually stated "not a valid conference number". This extension didn't do anything. Please help
rdelrosario
 
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Postby ramindia » Wed Jul 25, 2007 10:52 pm

Hi

this is your extension.conf dial plan issue

== Starting SIP/5933-081aaad0 at default,8600069,1 failed so falling back to exten 's'
== Starting SIP/5933-081aaad0 at default,s,1 still failed so falling back to context 'default'
Jul 25 18:21:37 WARNING[32695]: pbx.c:2377 __ast_pbx_run: Channel 'SIP/5933-081aaad0' sent into invalid extension 's' in context 'default', but no invalid handler


check again the dial plan

ram
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so its very usefull for others who join later as a NEWBIE.
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I was thinking that which why I posted it, but...

Postby rdelrosario » Wed Jul 25, 2007 11:36 pm

I don't understand what part of my extensions.conf its getting hung up on. I don't understand what those messages really mean. Can you explain to me what you think its doing incorrectly or explain to me what its trying to do and maybe isn't suppose to be doing?

The other extension as I posted seemed to get a different message. Not sure why one extension would get different messages. The other extension actually stated "That is not a valid conference".

It would really really help me greatly if someone could tell me what vicidial is passing to the extensions.conf so I know what part its trying to run... when it rings my phone, what command is being run in the extensions.conf. That part of it is working. Then the thing that confuses me most is what part of VICI and more specfiically what extensions are being passed to the extensions.conf after my phone rings. I see nothing but extensions starting with 86 and such. If I knew what VICI was trying to pass after it rings my phone maybe I could trace the dialplan, but I'm confused. any help would be great.
rdelrosario
 
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Postby ramindia » Thu Jul 26, 2007 1:39 am

Hi

Just look some example configs available at doc
in the source directory

ram
Kindly post your feedback, if this solution works.
so its very usefull for others who join later as a NEWBIE.
ramindia
 
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Just a summary would be helpful

Postby rdelrosario » Thu Jul 26, 2007 7:21 am

I undertand the samples for the most part, but I can't for the life of me find any documentation on how VICI dials the agent (the process) and how and what conference number it chooses to put the agent in and how the numbering scheme is chosen for conferencing and how VICI passes these seemingly arbitrary numbers to the dialplan for parsing? If you know, please just give me a bit of direction if you know the answer.

Thanks
rdelrosario
 
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Postby mflorell » Thu Jul 26, 2007 8:56 am

If you follow the SCRATCH_INSTALL document you will not get this message. What it means is that you either did not copy over the example meetme.conf file or you did not enter the meetme.conf room records properly.
mflorell
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Scratch-Install

Postby rdelrosario » Thu Jul 26, 2007 10:54 am

I literally typed in every line in the scratch-install and believe I didn't miss anything. I know I did something wrong in the mysql database as there are multiple sets of conference rooms. Are you saying the errors are something in the database or just in the meetme.conf file because I double checked the meetme.conf area as well as the extensions.conf are and they exist. If its in the database is there a way to rectify or fix the database issues without rebuilding from scratch?

Matt, thanks for replying, I'm trying really hard to get a handle on this and I'm pretty intuitive. Can you tell me what part of the extensions.conf file is run after my phone rings. I'm just trying to see where in the extensions.conf its trying to run, but not sure what VICI is passing to extensions to pattern match? This explanation might turn a light bulb in my head.

Lastly, I'm only have 2 sip phones setup with asterisk... no zaptel and more importantly no ZT-DUMMY that I've been reading about. This lack of ZT-Dummy wouldn't cause the above problems would it.

Thanks for your time... I really do appreciate it.
rdelrosario
 
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Postby aster1 » Thu Jul 26, 2007 11:21 am

Do you have any line like

exten => 8600069,1,Meetme(8600069) in you extensions.conf ? Do you have 8600069 in meetme.conf ?



did you install zaptel properly ?? Looking at your first set of warnings .. they are meetme and zaptel related .. the second one shows you have some error in extensions.conf . Even though you dont use only voip in asterisk the zaptel module is still very much required for meetme to function . Either you installed wrong version of zaptel with your asterisk or you dont have kernel modules loaded for meetme to function . Try
modprobe zaptel
modprobe ztdummy

Jul 25 18:23:39 WARNING[417]: chan_zap.c:1064 zt_open: Unable to open '/dev/zap/pseudo': No such device
Jul 25 18:23:39 ERROR[417]: chan_zap.c:7551 chandup: Unable to dup channel: No such device
Jul 25 18:23:39 WARNING[417]: app_meetme.c:464 build_conf: Unable to open pseudo channel - trying device
Jul 25 18:23:39 WARNING[417]: app_meetme.c:467 build_conf: Unable to open pseudo device
aster1
 
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Postby rdelrosario » Thu Jul 26, 2007 12:33 pm

I have the following related line in extensions.conf
exten => 8600069,1,Meetme,8600069

in my meetme.conf I do have the reference under [rooms] as specified in the scratch install.

The syntax above is much different than the one you mentioned in your reply... Did I do something wrong.

Thanks for replying.. I appreciate your time.
rdelrosario
 
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sorry more left off last response

Postby rdelrosario » Thu Jul 26, 2007 12:34 pm

Can I do the zaptel commands you mentioned without re-doing other asterisk things.... I might have skipped that section all together as I wasn't using zap... please advise.

Thanks
rdelrosario
 
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Postby aster1 » Thu Jul 26, 2007 1:55 pm

Here's some instruction :

Download zaptel tar.gz file from digium site .
cd in it and do a "make install"

Then go in you asterisk sources directory ( from where u installed previously ) and do a "make clean" and "make install" ( just in case this will recompile app-meetme as per new zaptel installed ) . This wont change anything in asterisk or vicidial configurations so you dont need to worry :) .
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modprobe ztdummy

Postby rdelrosario » Thu Jul 26, 2007 3:07 pm

I followed your instructions and when I run modprobe ztdummy it gives me some ushc error. modprobe zaptel worked fine. and I recompiled in the aterisk directory.. didn't check, but I know that ztdummy states its not available

root@phone:/usr/src/asterisk/asterisk-1.2.17# modprobe ztdummy
/lib/modules/2.4.33.3/kernel/drivers/usb/host/usb-uhci.o.gz: init_module: No such device
/lib/modules/2.4.33.3/kernel/drivers/usb/host/usb-uhci.o.gz: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.33.3/kernel/drivers/usb/host/usb-uhci.o.gz: insmod /lib/modules/2.4.33.3/kernel/drivers/usb/host/usb-uhci.o.gz failed
/lib/modules/2.4.33.3/kernel/drivers/usb/host/usb-uhci.o.gz: insmod ztdummy failed

Please advise. We are running 2.4.33 linux kernal running slackware (per the scratch-install)
rdelrosario
 
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2.6 kernal

Postby rdelrosario » Thu Jul 26, 2007 3:42 pm

In the scratch install I did the 2.4 kernal. Can I recompile to a 2.6 kernal by just following option 1 or will it completely break my SO SO working setup? Please advise

Thanks!
rdelrosario
 
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Postby aster1 » Fri Jul 27, 2007 12:23 am

I will really recommend you to use kernel 2.6 and recompile zaptel from new kernel . In kernel 2.4 zaptel depends on usb-uhci for timing and gives lot of problems .. Either get a X100p card for timing souce or recompile kernel 2.6 with timer frequency of 1000 HZ ( shown in scratch_install ) .
aster1
 
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Postby rdelrosario » Fri Jul 27, 2007 7:47 am

Yea, I just ordered a x100p card in hopes that this is my problem. In the mean time, I'm going to follow your advice and try to recompile the kernal to 2.6. Can I just follow the scratch install option 1 to RECOMPILE then follow your instructions above to recompile the zaptel stuff. then recompile asterisk again? Or is this not possible without breaking something else?

Thanks
rdelrosario
 
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Postby aster1 » Fri Jul 27, 2007 8:12 am

Yes .. follow step 1 in scratch_install to recompile kernel .. however do it this way only if you are familiar with recompiling kernel or else i would suggets you to get a 2.6 kernel package for your linux distribution and use that . When you haev booted in new kernel give a "make clean" and "make install" in zaptel directory and after that in asterisk source directory ..
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