[serusers] SER w/Asterisk IVR

Tracy Lofton tracylofton_1 at hotmail.com
Wed Sep 20 22:52:49 CEST 2006


Hi everyone,

I have integrated asterisk and openser which for the most part works great 
with this small exception that I am trying to get right.  My problem is,  I 
have users dial *86, *861, and *862 plus their extensions to access and 
utilize various different IVR of the Asterisk system which works great on 
SIP hardphones.  However, I am having a difficult time incorporating this 
feature into the Linksys Pap2t-NA unit.  I have changed it the .cfg file to 
accept just 861 and 862 for this unit, because the box does ont like the * 
(star dialed first), so i am trying to just use a 861 plus extension to 
retrieve voicemail for these units, but this is what I get from the Asterisk 
console:

Executing Answer("SIP/31002-09d67ec8", "") in new stack
    -- Executing VoiceMailMain("SIP/31002-09d67ec8", "") in new stack
    -- Playing 'vm-login' (language 'en')
Sep 20 13:24:06 WARNING[13992]: app_voicemail.c:5002 vm_authenticate: 
Couldn't read username
    -- Executing Answer("SIP/31002-09d85c30", "") in new stack
    -- Executing VoiceMailMain("SIP/31002-09d85c30", "") in new stack
    -- Playing 'vm-login' (language 'en')
    -- Playing 'vm-password' (language 'en')
Sep 20 13:24:45 WARNING[13998]: app_voicemail.c:5034 vm_authenticate: Unable 
to read password
    -- Executing Ringing("SIP/31002-09d82da0", "") in new stack
    -- Executing VoiceMail("SIP/31002-09d82da0", "u31002") in new stack
    -- Playing 'vm-theperson' (language 'en')
    -- Playing 'digits/3' (language 'en')
    -- Playing 'digits/1' (language 'en')
    -- Playing 'digits/0' (language 'en')
    -- Playing 'digits/0' (language 'en')
    -- Playing 'digits/2' (language 'en')
    -- Playing 'vm-isunavail' (language 'en')
  == Spawn extension (default, 131002, 2) exited non-zero on 
'SIP/31002-09d82da0'

Should the spawn portion be displaying the extra "1" in front of the 
extension 31002?  Also, is there a special area to input a password for the 
Pap2t via the web interface so that it is readable by Asterisk IVR system, 
maybe a callerID or...?  Or is there some other way, I can work around this 
just for Linksys Pap2t users (and other ATA units) that someone is familiar 
with already?

TIA.

Tracy

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