On 9/21/06, Tracy Lofton <tracylofton_1@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@hotmail.com>
>To: serusers@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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