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Not a valid conference number

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Not a valid conference number

Postby joschi05fan » Fri Jul 14, 2006 4:19 am

hi @ all,

i've got vicidial finally to work, now there is just one last problem to solve. maybe someone can help me with that.
after login in vicidial my sip phone rings, and if i pick up the phone a nice womans voice says:
Code: Select all
that is not a valid conference number, please try again

and in the cli the following message appears
Code: Select all
VD_hangup : 102  SIP/102-2ef8 2
+++++ VD hangup START : |1152865477.0|SIP/102-2ef8|h|SIP|102|0|2006-07-14 10:24:39||2|


|SELECT lead_id,callerid FROM vicidial_auto_calls where uniqueid = '1152865477.0' limit 1;|

VD hangup: no VDAC record found: 1152865477.0
Jul 14 11:12:54 WARNING[28609]: channel.c:2538 ast_request: No channel type registered for 'zap'
Jul 14 11:12:54 WARNING[28609]: app_meetme.c:460 build_conf: Unable to open pseudo channel - trying device
Jul 14 11:12:54 WARNING[28609]: app_meetme.c:463 build_conf: Unable to open pseudo device
+++++ CALL LOG HUNGUP: |1152868370.2|SIP/102-1070|h|2006-07-14 11:12:57|min: |
Perl Environment Dump:
0|PRI-----NODEBUG-----16---------------|
AGI Environment Dump:
 -- accountcode =
 -- callerid = unknown
 -- calleridname = S0607141112508600051
 -- callingani2 = 0
 -- callingpres = 0
 -- callingtns = 0
 -- callington = 0
 -- channel = SIP/102-1070
 -- context = default
 -- dnid = unknown
 -- enhanced = 0.0
 -- extension = h
 -- language = en
 -- priority = 2
 -- rdnis = unknown
 -- request = VD_hangup.agi
 -- type = SIP
 -- uniqueid = 1152868370.2
AGI Environment Dump: |1152868370.2|SIP/102-1070|h|SIP|S0607141112508600051|S0607141112508600051|2|

DEBUG: NODEBUG

VD_hangup : S0607141112508600051  SIP/102-1070 2
+++++ VD hangup START : |1152868370.2|SIP/102-1070|h|SIP|S0607141112508600051|508600051|2006-07-14 11:12:58||2|S0607141112508600051|


|SELECT lead_id,callerid FROM vicidial_auto_calls where uniqueid = '1152868370.2' limit 1;|

VD hangup: no VDAC record found: 1152868370.2 S0607141112508600051
Thx

Joschi
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if you don't know the direction, then no wind will fit for you
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Location: Cloppenburg, Germany
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Postby mflorell » Fri Jul 14, 2006 7:06 am

What is your zaptel timing device?

Do you have a zaptel card installed on your system?

Are you using ztdummy? if so what Linux kernel and what timing source are you using?
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Postby joschi05fan » Fri Jul 14, 2006 7:14 am

hi,

i don't have a zaptel device. i'm tried to install ztdummy.
i'm using slackware linux with kernel 2.4.3.1 like it is recomended in the scratch install.
Thx

Joschi
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Location: Cloppenburg, Germany
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Postby mflorell » Fri Jul 14, 2006 9:28 am

I would recommend either buying a cheap x100p card($20-$40) or trying out the experimental app_conference module which doesn't require a zaptel timer.
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i got it

Postby joschi05fan » Mon Jul 17, 2006 1:47 am

we didn't want to use a isdn card, so i configured ztdummy, and now it works.

joschi
Thx

Joschi
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if you don't know the direction, then no wind will fit for you
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Joined: Fri Jul 14, 2006 4:07 am
Location: Cloppenburg, Germany
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well posted to fast

Postby joschi05fan » Mon Jul 17, 2006 7:45 am

now the dial goes out, but i do not get the Lead Disposition Screen. Only a blank green screen.

any suggestions?
Thx

Joschi
_________________________________________________
if you don't know the direction, then no wind will fit for you
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Joined: Fri Jul 14, 2006 4:07 am
Location: Cloppenburg, Germany
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Postby mflorell » Mon Jul 17, 2006 9:54 am

Please post some Real Asterisk CLI output.

Also, look in your Javascript Console in Firefox and see if any errors are showing up there when you login to vicidial.php
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Postby joschi05fan » Tue Jul 18, 2006 5:09 am

here is the code, that came on the asterisk - cli

Code: Select all
VD_hangup : 103  SIP/103-997f 2
+++++ VD hangup START : |asterisk-3166-1153204724.0|SIP/103-997f|h|SIP|103|0|2006-07-18 8:38:56||2|anonymous|


|SELECT lead_id,callerid FROM vicidial_auto_calls where uniqueid = 'asterisk-3166-1153204724.0' limit 1;|

VD hangup: no VDAC record found: asterisk-3166-1153204724.0 anonymous
Jul 18 08:44:29 WARNING[3651]: channel.c:2570 ast_request: No channel type registered for 'zap'
Jul 18 08:44:29 WARNING[3651]: app_meetme.c:460 build_conf: Unable to open pseudo channel - trying device
Jul 18 08:44:29 WARNING[3651]: app_meetme.c:463 build_conf: Unable to open pseudo device
+++++ CALL LOG HUNGUP: |asterisk-3166-1153205065.2|SIP/103-2576|h|2006-07-18 8:44:33|min: |
Perl Environment Dump:
0|PRI-----NODEBUG-----16---------------|
AGI Environment Dump:
 -- accountcode =
 -- callerid = unknown
 -- calleridname = S0607180844248600051
 -- callingani2 = 0
 -- callingpres = 0
 -- callingtns = 0
 -- callington = 0
 -- channel = SIP/103-2576
 -- context = default
 -- dnid = unknown
 -- enhanced = 0.0
 -- extension = h
 -- language = en
 -- priority = 2
 -- rdnis = unknown
 -- request = VD_hangup.agi
 -- type = SIP
 -- uniqueid = asterisk-3166-1153205065.2
AGI Environment Dump: |asterisk-3166-1153205065.2|SIP/103-2576|h|SIP|S0607180844248600051|S0607180844248600051|2|

DEBUG: NODEBUG

VD_hangup : S0607180844248600051  SIP/103-2576 2
+++++ VD hangup START : |asterisk-3166-1153205065.2|SIP/103-2576|h|SIP|S0607180844248600051|248600051|2006-07-18 8:44:33||2|S0607180844248600051|


|SELECT lead_id,callerid FROM vicidial_auto_calls where uniqueid = 'asterisk-3166-1153205065.2' limit 1;|

VD hangup: no VDAC record found: asterisk-3166-1153205065.2 S0607180844248600051
Jul 18 08:44:44 WARNING[3692]: channel.c:2570 ast_request: No channel type registered for 'zap'
Jul 18 08:44:44 WARNING[3692]: app_meetme.c:460 build_conf: Unable to open pseudo channel - trying device
Jul 18 08:44:44 WARNING[3692]: app_meetme.c:463 build_conf: Unable to open pseudo device
Jul 18 08:44:45 WARNING[3692]: file.c:1035 ast_waitstream: Unexpected control subclass '14'
+++++ CALL LOG HUNGUP: |asterisk-3166-1153205084.6|Local/8600051@default-cf93,2|h|2006-07-18 8:44:48|min: |
Perl Environment Dump:
0|PRI-----NODEBUG-----0---------------|
AGI Environment Dump:
 -- accountcode =
 -- callerid = 4944716666
 -- calleridname = M0718084443000000667
 -- callingani2 = 0
 -- callingpres = 0
 -- callingtns = 0
 -- callington = 0
 -- channel = Local/8600051@default-cf93,2
 -- context = default
 -- dnid = unknown
 -- enhanced = 0.0
 -- extension = h
 -- language = en
 -- priority = 2
 -- rdnis = unknown
 -- request = VD_hangup.agi
 -- type = Local
 -- uniqueid = asterisk-3166-1153205084.6
AGI Environment Dump: |asterisk-3166-1153205084.6|Local/8600051@default-cf93,2|h|Local|M0718084443000000667|M0718084443000000667|2|

DEBUG: NODEBUG

VD_hangup : M0718084443000000667  Local/8600051@default-cf93,2 2
+++++ VD hangup START : |asterisk-3166-1153205084.6|Local/8600051@default-cf93,2|h|Local|M0718084443000000667|667|2006-07-18 8:44:48||2|M0718084443000000667|--    VDhangup Local DEBUG: |PRI|M0718084443000000667||0|
+++++ VDAD START LOCAL CHANNEL: EXITING- 2
+++++ CALL LOG HUNGUP: |asterisk-3166-1153205084.4|Local/8600051@default-cf93,1|h|2006-07-18 8:44:48|min: |
Perl Environment Dump:
0|PRI-----NODEBUG-----0-----CANCEL----------|
AGI Environment Dump:
 -- accountcode =
 -- callerid = 4944716666
 -- calleridname = M0718084443000000667
 -- callingani2 = 0
 -- callingpres = 0
 -- callingtns = 0
 -- callington = 0
 -- channel = Local/8600051@default-cf93,1
 -- context = default
 -- dnid = unknown
 -- enhanced = 0.0
 -- extension = h
 -- language = en
 -- priority = 2
 -- rdnis = unknown
 -- request = VD_hangup.agi
 -- type = Local
 -- uniqueid = asterisk-3166-1153205084.4
AGI Environment Dump: |asterisk-3166-1153205084.4|Local/8600051@default-cf93,1|h|Local|M0718084443000000667|M0718084443000000667|2|

DEBUG: NODEBUG

VD_hangup : M0718084443000000667  Local/8600051@default-cf93,1 2
+++++ VD hangup START : |asterisk-3166-1153205084.4|Local/8600051@default-cf93,1|h|Local|M0718084443000000667|667|2006-07-18 8:44:48||2|M0718084443000000667|--    VDhangup Local DEBUG: |PRI|M0718084443000000667|CANCEL|0|
+++++ VDAD START LOCAL CHANNEL: EXITING- 2

*CLI>
Thx

Joschi
_________________________________________________
if you don't know the direction, then no wind will fit for you
joschi05fan
 
Posts: 8
Joined: Fri Jul 14, 2006 4:07 am
Location: Cloppenburg, Germany
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Postby mflorell » Tue Jul 18, 2006 8:06 am

You are still having issues with your zaptel timer:
"Unable to open pseudo channel"

meetme will not work until that is fixed.
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Postby joschi05fan » Tue Jul 18, 2006 8:10 am

mhh i thought that you would write this, after i read the post... :lol:
can you help me with that???
i already reinstallt asterisk, but i still get the error.
Thx

Joschi
_________________________________________________
if you don't know the direction, then no wind will fit for you
joschi05fan
 
Posts: 8
Joined: Fri Jul 14, 2006 4:07 am
Location: Cloppenburg, Germany
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Postby mflorell » Tue Jul 18, 2006 8:33 am

I'm not that good at getting ztdummy to work. Did you modify your code to compile ztdummy properly?

Have you looked at the wiki?
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
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Postby jansaell » Tue Jul 18, 2006 1:32 pm

test after you have loaded the modules that they are in there by doing an lsmod and make shure that you can se the modules:
Code: Select all
Module                  Size  Used by
ztdummy                 7744  0
zaptel                240644  147 ztdummy
crc_ccitt               6144  1 zaptel
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well finally..

Postby joschi05fan » Wed Jul 19, 2006 4:26 am

... it's now working...
i reinstalled ztdummy and zaptel.
the blank screen we fixed by reloading the sql tables. the maintables where empty, so we filled it. now it is working...
thanx for all the help, and keep up with the good work.
Thx

Joschi
_________________________________________________
if you don't know the direction, then no wind will fit for you
joschi05fan
 
Posts: 8
Joined: Fri Jul 14, 2006 4:07 am
Location: Cloppenburg, Germany
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Postby jlodvo » Mon Aug 14, 2006 7:31 am

hello

can anyone help me out im at installing asterisk and done 1.2 release next is im getting a hardtime compiling ztdummy can anyone help me or make a scratch_install for it? it would rilly be a great help thanks
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