[kspare]

I can't find this in the manual, maybe I missed it.

How does a user call into the system from an external line to pick up their voicemail?

I am using pstn lines for my phone system, no sip lines at all. Right now i've got my system setup so the aa answers all calls. But i'm not sure from there how to get into a users vm?

Help!

[humba3]

That topic has come up before Here's a little hint: you create a "virtual" extension which results in a call to the VM pickup number (800@ip-spa400:5090). If you need more details, the forum's search engine will come in handy.

[kspare]

It's not really popping up for me. Could you explain how to do this? maybe the mods can make this a sticky for others to find easily?

[kspare]

Finally found the solution.

This is now my AA dial plan.

( 1| <2:500>S0 | <3:501>S0 | <#:800@spa400-ip:5090>S0 )

User dials 1, gets our hours, dials 2 for hunt group 1, dials 3 for huntgroup 2, hits # (pound) for vm. works perfectly!!

http://forum.voxilla.com/linksys-sip...html#post92676

[Juarez]

Kspare,
I've been trying tp do this with no luck.
the XML I tried is posted below.
Can you give me some feedback to make this work?

Objective:
dial 1 for location
dial 2 for directory
dial * for VM

Mike.


<aa>
<form id="dt" type="menu">
<audio src="prompt5" bargein="T"/>
<noinput timeout="10" repeat="T"/>
<nomatch repeat="F">
<audio src="prompt3" bargein="T"/>
</nomatch>
<dialplan src="dp1"/>
<match>
<case input="1">
<audio src="prompt7" bargein="F"/>
</case>

<case input="2">
<audio src="prompt8" bargein="F"/>
</case>
<default>
<audio src="prompt2"/>
<xfer name="ext" target="$input"/>
</default>
</match>
</form>
</aa>

[kspare]

what exactly isn't working and what is?

[Juarez]

The second <match> case does not seem to register.
If you select 2, the system just seems to hang.

M.

[kspare]

I'll try and take a look in the manual tonight and see what I can find on that.

[Brad_Delta360]

I had some trouble setting this up as well. What I noticed was that my transfer to the SPA400 was being truncated. I had <*:800@192.168.1.241:5090> and what the syslogs where showing was a transfer to 800@192.168.1.241:50 so it was droping the last two digits. I changed the extention from 800 to 8 and now it works. Any one else have that problem? Is there a size limit I am not aware of?

Brad..