asterisk users - Dec 2006 - sip address in voicemail emails

hi,

I am using asterisk 1.2.10.
I am trying to send sip links in asterisk voicemail, so that users can
easily reply to emails.
This does not seem to be straightforward.
First, there seems to be no variable that prints out the domain name of the
sip call, since I am including every variable mentioned on
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
Second, if i include in the emailbody variable sip:${VM_CIDNUM}@mydomain (on
the wrong assumption that the phone call was from a telephone number), gmail
responds by assuming it is a mailto link, and outlook treats it as plain
text.  Having examined the emails being sent in mutt, it appears that the
message has no mime type (neither text nor text/html).  Unsurprisingly,
then, enclosing it in an href, i.e. <a href=sip:my@address.com>click
here<a/> is of no help.

Is there  a workaround for this?

Perhaps it is being addressed in a later version of asterisk?  I have seen
no mention of it in the release notes for 1.4.

Thanks
Mark Price
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Hi,

Item zero, Thanks for hosting and participating in this list, digium, and
all the developers involved.

First, I realize that my first post probably did not belong on
asterisk-dev.  I had intended to remove that address from my recipients
list, but did not
Secondly,  I figure that now that my neck is extended, I should write to
clarify one thing and correct another in my first email.

On 12/1/06, Mark Price <markprice@gmail.com>
wrote:
>
> hi,
>
> I am using asterisk 1.2.10.
> I am trying to send sip links in asterisk voicemail, so that users can
> easily reply to emails.
> This does not seem to be straightforward.
> First, there seems to be no variable that prints out the domain name of
> the sip call, since I am including every variable mentioned on
> http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf

To clarify to myself, I made a call from a different sip domain from a
username that does not exist on the asterisk box, and found out that it is
true: VM_CIDNUM contains the username, but not the domain name of the call.
Therefore, as long as the username is a telephone number, we can work around
that, but the message printed to describe a non-telephone-number phone call
will be incorrect.

Second, if i include in the emailbody variable sip:${VM_CIDNUM}@mydomain
(on
> the wrong assumption that the phone call was from a telephone number),
gmail
> responds by assuming it is a mailto link, and outlook treats it as plain
> text.  Having examined the emails being sent in mutt, it appears that the
> message has no mime type (neither text nor text/html).

I thought I should point out I was incorrect.  The text portion of the email
is given mime-type "text."  Therefore it appears to be impossible to
send a
sip link in the email unless the receiving email client knows how to
recognize them (as many know how to recognize http:// and mailto: links).

Thanks,
Mark Price
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Am Freitag, den 01.12.2006, 17:57 -0500 schrieb Mark
Price:
> Hi, 
> On 12/1/06, Mark Price <markprice@gmail.com> wrote:
>         hi,
>         
>         I am using asterisk 1.2.10.
>         I am trying to send sip links in asterisk voicemail, so that
>         users can easily reply to emails.
>         This does not seem to be straightforward.
>         First, there seems to be no variable that prints out the
>         domain name of the sip call, since I am including every
>         variable mentioned on http://www.voip-info.org/wiki-Asterisk
>         +config+voicemail.conf
You might use your own notification script, s.t. you can send a
Content-type: text/html for your E-Mail.
> To clarify to myself, I made a call from a different sip domain from a
> username that does not exist on the asterisk box, and found out that
> it is true: VM_CIDNUM contains the username, but not the domain name
> of the call.  Therefore, as long as the username is a telephone
> number, we can work around that, but the message printed to describe a
> non-telephone-number phone call will be incorrect. 
Hi Mark,

I guess there are incoming "lines" that send callerid rather reliably
-
PSTN will most probably send the correct number, if it sends one. OTOH,
if you accept incoming SIP connections from unauthenticated hosts, they
could basically send any "from" phone number they wish. Your dialplan
should probably take care of that, possibly by prepending the callerid
with some prefix ("X_" or whatever). Your notification script could
make
use of that information to _not_ provide a "callback link" in that
case.

Several programs allow for special protocol handling; which mail
client/internet browser you use will determine wether you can configure
it to hand off "sip:" URIs to the proper program (X-Lite, a CTI
prog...).

BR
Anselm