[asterisk-users] Polycom 601 and Paging

[Bruce Reeves]

Do you have an enrty like this? Or do you have both a sip.cfg and an
ipmid.cfg file?

<alertInfo voIpProt.SIP.alertInfo.2.value="RA"
voIpProt.SIP.alertInfo.2.class="4"/>

On 7/18/06, Dovid Bender <asteriskusers at dovid.net> 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)" <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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-- 
Bruce
Nortex Networks
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