20 Sep
2006
20 Sep
'06
10:09 p.m.
The Asterisk/Asterisk list might get your problem fixed faster.
On 9/21/06, Tracy Lofton <tracylofton_1(a)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(a)hotmail.com>
To: serusers(a)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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