[asterisk-users] Polycom 601 and Paging

Tue Jul 18 14:36:49 MST 2006

Are you using the Polycom config files from krisk.org mentioned on the
wiki?  They're pretty good.  

I would think you'd need this in your sip.cfg at the very least.  Maybe
put this at the bottom of the file:

<sip><voIpProt>
  <alertInfo voIpProt.SIP.alertInfo.1.value="AA"
voIpProt.SIP.alertInfo.1.class="3"/>
  <alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer"
voIpProt.SIP.alertInfo.2.class="4"/>
</voIpProt></sip>

I just realized I referenced the tags backwards in my post on the wiki.

-------------------
Brian Vincent
Copper Mountain Telecom
vincentb at coppercolorado.com

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dovid
Bender
Sent: Tuesday, July 18, 2006 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 601 and Paging

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)" <VincentB at coppercolorado.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
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
vincentb at coppercolorado.com

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