My sip.cfg (v.1.6.5) has the following lines:
<alertInfo voIpProt.SIP.alertInfo.1.value=""
voIpProt.SIP.alertInfo.1.class=""/>
There is not a "2" version.
and
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer"
se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6"
se.rt.4.mod="1"/>
At 04:27 PM 7/18/2006, you wrote:
I cant do step 2.
I cant find:
2. Okay, see how the SIPAddHeader includes "Ring Answer"? That word
or words will be matched by alertInfo in sip.cfg in order to figure
out what to do. You are using the config files from krisk.org listed
above, right? If not, go get them now. I'll wait. So in sip.cfg in
the <voIpProt><SIP> section you need a line like:
<alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer"
voIpProt.SIP.alertInfo.2.class="4"/>
----- Original Message ----- From: "Brian Vincent (C)"
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Sent: Tuesday, July 18, 2006 5:09 PM
Subject: RE: [asterisk-users] Polycom 601 and Paging
I have these instructions on the wiki in the comments section. I
had a hard time following the directions too, but I finally got it to work:
We've got 3 things going on with setting up Auto Answer and Ring
Answer. Let's detail this process from beginning to end using Ring
Answer as our example. (Auto Answer isn't much different except you
want to make sure step #2 below goes to class 3 rather than 4, and
that class 3 is set up as described elsewhere which is the same as
the one in the ipmid.cfg file from krisk.org.)
1. First, use the SIPAddHeader() directive in Asterisk to properly
alert the phone. In my situation, I have 10 phones with 2-digit
extensions. I want to call each phone by prefixing the extension
with a "1" in order to activate the intercom. For example, if I dial
126 I want it to put extension 26 on speakerphone. So go into
extensions.conf and make sure you create a new section like this:
<a href='icm-auto-answer'>icm-auto-answer </a href='icm-auto-answer'>
;intercom
exten => _12x,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => _12x,2,Dial(sip/${EXTEN:1:3})
exten => _12x,3,Hangup
exten => _12x,102,Hangup
Then make sure in your from-internal section of extensions.conf you
have a include => icm-auto-answer
2. Okay, see how the SIPAddHeader includes "Ring Answer"? That word
or words will be matched by alertInfo in sip.cfg in order to figure
out what to do. You are using the config files from krisk.org listed
above, right? If not, go get them now. I'll wait. So in sip.cfg in
the <voIpProt><SIP> section you need a line like:
<alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer"
voIpProt.SIP.alertInfo.2.class="4"/>
The value parameter must match whatever you use in the SIPAddHeader
string. In this case they're both "Ring Answer". You could just as
easily replace both with the word "Foo" or "RA".
3. Now, the alertInfo tag will match that value and then go to the
"class" value to figure out what to do. Se we need to make sure
class="4" is set up properly. You could probably set up class 4 in
sip.cfg, but mine lives in ipmid.cfg. So go into ipmid.cfg and
locate the <ringtypes> section. Below that tag (and before it's
corresponding </ringtype> closing tag) you need to make sure class 4
is set up right. You should have this line:
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer"
se.rt.4.timeout="1000" se.rt.4.ringer="2" se.rt.4.callWait="6"
se.rt.4.mod="1"/>
The notes above describe that line. The key is that this is class 4
as noted by the 3rd part of the value names - se.rt.4.name. I'd like
to add that the keyword "RING_ANSWER" is meaningless, it's just a
human-readable tag.
Got all that? The SIPAddHeader of "Ring Answer" hits the <alertInfo>
tag to figure out which class to go to. Then the class in ipmid.cfg
says, "Oh, I'm a "ring-answer" type and my firmware knows what to do
with that type."
One test you can do is to connect to asterisk ($ asterisk -r), bump
your verbosity up (<tt>set verbose 6</tt>), and try to place a call
using that context from step #1. You'll see one phone calling
another and within the Asterisk CLI you should see the following
message appear:
- Executing SIPAddHeader("SIP/20-86bc", "Alert-Info: Ring Answer")
in new stack<br />
Extension Changed 20 new state InUse for Notify User 26<br />
- Executing Dial("SIP/20-86bc", "sip/26") in new stack<br />
- Called 26<br />
- SIP/26-0448 is ringing<br />
- SIP/26-0448 answered SIP/20-86bc<br />
- Attempting native bridge of SIP/20-86bc and SIP/26-0448
If you don't see that Alert-Info: Ring Answer being sent, then you
know you haven't gotten the first step right.
Also, I made the mistake of putting some comments into the .cfg
files and the comments seemed to screw up the parser. It ignored
seemingly random lines (i.e. non-comment ones). I'm not a complete
moron since I've been writing XML for 6 years (and HTML for 11) but
it goes to show how careful you should be. Anyway, I use "xmllint"
on config files now before rebooting the phones to make sure I
didn't make a dumb typo.
-------------------
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED]
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