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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: soprom on October 07, 2006, 09:53:47 PM
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When recording greeting, the sequence goes like this:
1. I dial *60*1002 (greeting id)
pbx says "enter password" (from the general settings?)
2. I say my thing and end with #
pbx says "123123" and seems to wait for a valid extension
Any valid extension entered at this point is rejected.
Where am I going wrong?
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Hi Sophie
Password is password for keyops specified in globals. Default is 1111.
After correct password, system will confirm the greeting number it is about to record followed by recording instructions and then a beep when it begins recording. Press # after recording your greeting. System will play back your greeting and then give you the option to save your message or re-record it.
Suspect you are entering wrong password and I also suspect a bug in the code when incorrect password is entered. I will look at this tomorrow when I can get to system. In the meantime it should work fine if you have the correct password.
Kind Regards
Selintra
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There is no beep before recording. I have to speak and end it with #.
There is no indication to save or not...
So I'll wait for you then! Many thanks for your assistance.
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It appears that you may not have the full asterisk soundpack fitted
do....
rpm -q smeserver-asterisk-sounds
You should have 1.2.2 fitted. If not, you can get it here
http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/
If you do have them fitted, could you post a verbose console log when you do your record? We have customers recording greetings every day without problems so I think this may be a simple problem to fix.
Kind Regards
Selintra
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Thanks, it's working.
Since I use the system in french, I didn't bother installing sounds.
But the sounds package is requiered!
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Hi Sophie
There is a French sounds pack in the language pack library. It's French rather than quebecois but it may be of some use to you. It's here...
http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/languagepacks/
Kind Regards
Selintra
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I know but it's not suitable for Québec...
For Québec I use http://newton.waglo.com/~millette/asterisk/ and copy all files in /var/lib/asterisk/sounds
PS
Thanks for caring!
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Password is password for keyops specified in globals. Default is 1111.
After correct password, system will confirm the greeting number it is about to record followed by recording instructions and then a beep when it begins recording. Press # after recording your greeting. System will play back your greeting and then give you the option to save your message or re-record it.
Suspect you are entering wrong password and I also suspect a bug in the code when incorrect password is entered. I will look at this tomorrow when I can get to system. In the meantime it should work fine if you have the correct password.
Kind Regards
Selintra
Mine says :"password incorrect, please enter your password followed by the pound key"
The password specified in globals for keyops is "1111", but its just not working, and it used to work. what now selintra?
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Works fine on our test rig and this is a heavily used area of the system so I don't expect too many outstanding bugs here. I would suspect the DTMF settings, wither in your phone or in sip.conf. That's where I would check first. Also, watch the console log when you are inputting your numbers, run the system in agi debug and you will see what pasword the system is expecting ... Here's an example...
-- Executing [*60*2345@internal:1] BackGround("SIP/4020-082d69e8", "silence/1") in new stack
-- <SIP/4020-082d69e8> Playing 'silence/1' (language 'en')
-- Executing [*60*2345@internal:2] AGI("SIP/4020-082d69e8", "selintra|*60*2345") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
AGI Tx >> agi_request: selintra
AGI Tx >> agi_channel: SIP/4020-082d69e8
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1225227684.185
AGI Tx >> agi_callerid: 4020
AGI Tx >> agi_calleridname: Snom
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: *60*2345
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: internal
AGI Tx >> agi_extension: *60*2345
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << EXEC Authenticate 1111
-- AGI Script Executing Application: (Authenticate) Options: (1111)
-- <SIP/4020-082d69e8> Playing 'agent-pass' (language 'en')
-- <SIP/4020-082d69e8> Playing 'auth-thankyou' (language 'en')
Regards
S
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Thanx for the fast reply.
It was working fine and now its not, here's my output:
[root@pabx ~]# asterisk -rvvvvv
Asterisk 1.4.21.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.21.1 currently running on pabx (pid = 7191)
Verbosity is at least 5
pabx*CLI> agi debug
AGI Debugging Enabled
-- Executing [*60*0000@internal:1] BackGround("SIP/5002-09d5b770", "silence/1") in new stack
-- <SIP/5002-09d5b770> Playing 'silence/1' (language 'en')
-- Executing [*60*0000@internal:2] AGI("SIP/5002-09d5b770", "selintra|*60*0000") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
AGI Tx >> agi_request: selintra
AGI Tx >> agi_channel: SIP/5002-09d5b770
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1225228820.1
AGI Tx >> agi_callerid: 102
AGI Tx >> agi_calleridname: Johan
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: *60*0000
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: internal
AGI Tx >> agi_extension: *60*0000
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << EXEC Authenticate 1111
-- AGI Script Executing Application: (Authenticate) Options: (1111)
-- <SIP/5002-09d5b770> Playing 'agent-pass' (language 'en')
-- <SIP/5002-09d5b770> Playing 'auth-incorrect' (language 'en')
as far as i know the only thing i changed was to add:
limitonpeers=yes
notifyringing=yes
notifybusy=yes
in sip.conf header and
call-limit=99
for each phone to get BLF working. it seems like its expecting the right password, wich is "1111". everything else is working fine.
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Ok, it seems like its working on all the extentions except the 2 ive been testing it on the whole night. Ive got about 14 extentions with budgetone-200 phones and with those it working. Then ive got a grandstream gxp-2000 and a grand stream analog telephone adapter with a corless phone connected to it, on wich it is not working. (i did do a factory reset on the gxp-2000). as long as i can record a greeting somewhere it is fine. but i would still like to know why it is not working. Thanx for your help so far. It is appreciated.
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Usually when a phone is apparently not sending DTMF, it is down to a mismatch between the kind of DTMF the phone is sending and the kind of DTMF which Asterisk is listening for. The phone DTMF is specified in the phone browser or in the provisioning file you use to set the phone up. It varies from phone type to phone type. Asterisk DTMF is specified in the sip.conf section for the phone (in SAIL, it appears in the left-hand window of the extension->edit browser). The asterisk keyword is dtmfmode – you can have:-
dtmfmode=info
dtmfmode=rfc2833
dtmfmode=inband
dtmfmode=auto.
The asterisk default is I believe inband
In theory, auto is a kind of catch all which will figure out for itself what’s going on but I’ve never had much success with it.
Kind Regards
S