[serusers] SER w/Asterisk IVR

[Weiter Leiter]

The Asterisk/Asterisk list might get your problem fixed faster.

On 9/21/06, Tracy Lofton <tracylofton_1 at hotmail.com> wrote:
>
> Hi all,
>
> I think I am making some progress, but need still need help.  Also, just
> to
> let you all of you know I am not double posting.  I chose to post this
> particular issue on SER mail list because there seem to be more
> SER/Asterisk
> people then OpenSER/Asterisk in my humble opinion.
>
> Well, in sip.conf, I added to one of the users of the ATA unit to solicit
> a
> change:
>
> dtmfmode=rfc2833
>
> which got rid of vm authentication problem.  However,  I now get this:
>
> e EXIT or QUIT to exit the asterisk console
>    -- Executing Ringing("SIP/31002-087e5d88", "") in new stack
>    -- Executing VoiceMail("SIP/31002-087e5d88", "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')
>    -- Playing 'vm-intro' (language 'en')
> == Spawn extension (default, 131002, 2) exited non-zero on
> 'SIP/31002-087e5d88'
>
> Just hangs me up completely.
>
>    -- Executing Answer("SIP/31002-088015e0", "") in new stack
>    -- Executing VoiceMailMain("SIP/31002-088015e0", "") in new stack
>    -- Playing 'vm-login' (language 'en')
>    -- Playing 'vm-password' (language 'en')
>    -- Incorrect password '' for user '31002' (context = default)
>    -- Playing 'vm-incorrect-mailbox' (language 'en')
>
> Just cuts me off with incorrect password and vm-incorrect-mailbox.  But
> the
> red error is no longer showing, but still not right.
>
> Thanks all, for any help you can provide.
>
> Tracy
>
>
> >From: "Tracy Lofton" <tracylofton_1 at hotmail.com>
> >To: serusers at iptel.org
> >Subject: [serusers] SER w/Asterisk IVR
> >Date: Wed, 20 Sep 2006 20:52:49 +0000
> >
> >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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