I feel your pain<br><br><div class="gmail_quote">On Fri, Mar 4, 2011 at 9:29 AM, Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><div></div><div class="h5">-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Louis Carreiro<br>
Sent: Friday, March 04, 2011 8:07 AM<br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: [asterisk-users] Asterisk <-> Lync / Call Center Transfer / Refer<br>
<br>
Hey all,<br>
<br>
Alright. So we decided to not go with Avaya for our next PBX and we are now<br>
full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our<br>
SIP gateway and call center and Lync is our internal UC and IP-PBX server.<br>
I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig<br>
and all is beautiful (except for the Opt11 not receiving names from * but<br>
that's another topic). So, my problem now is with the call center.<br>
<br>
This setup may be a bit convoluted at first but it'll make sense I hope.<br>
I've created the queues in Asterisk via FreePBX. I then created a ring group<br>
for each Lync extension so we get the "Confirm Calls" option and dodge the<br>
voice mail problem. The agents the login via their Lync phone with the Ring<br>
Group extension as their Agent ID. It kind of looks like this:<br>
<br>
Queue 2001<br>
Agent 4001<br>
Agent 4002<br>
Agent 4003<br>
<br>
Ring Group 4001 -> Lync Extention 5001<br>
Ring Group 4002 -> Lync Extention 5002<br>
Ring Group 4003 -> Lync Extention 5003<br>
<br>
This all works beautifuly! The problem I have is on transfers. If Lync<br>
extension 5001 trasnfers to Lync extension 5010, Asterisk is unaware of the<br>
transfer and shows that 5001 is still active with the call. We're using<br>
OrderlyStats to monitor the queue so I watch the "Talking" counter just keep<br>
counting instead of being aware the transfer took place. Now to me, that<br>
says to me that the transfer took place within Lync so Asterisk is unaware<br>
of the transfer. So my next step was to enable Refer support in Lync so Lync<br>
sends the refer message back to Asterisk to transfer the call so Asterisk is<br>
fully aware of what's going on. It seems like the refer message is trying to<br>
work and Lync is sending it and Asterisk is receiving it but the "Refer-To"<br>
is changing between the two so I'm at a loss.<br>
<br>
(Logs are below signature)<br>
Lync says it's sending the following message with a "Refer-to:<br>
<<a href="mailto:sip%3Auser2@domainname.com">sip:user2@domainname.com</a>>"<br>
<br>
Asterisk is seeing the following and the refer-to changed, it's now<br>
"REFER-TO:<br>
<sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad278<br>
7?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto<br>
-tag%3D8be38bb187>".<br>
<br>
At first it seems like Lync is sending a true SIP URI so I need to get<br>
Asterisk to know how to handle that SIP URI and then secondly, it seems like<br>
Asterisk doesn't even receive the same REFER-TO message that Lync sent. Is<br>
this because Asterisk doesn't know how to handle the SIP URI?<br>
<br>
So I guess I'm left with wondering if fixing the REFER message stuff is<br>
going to fix my problem even? The end goal is for Asterisk to be aware that<br>
a call was transferred to another extension in Lync.<br>
<br>
<br>
<br>
Thanks in advance everyone!<br>
Louis<br>
<br>
</div></div><snip><br>
<br>
First of all, I assume you are using 1.8.X. Regardless, Queueing and<br>
referring have some known issues. If you look at chan_sip.c, you'll see<br>
that REFER is considered "broken" at this time (I know this to be the case<br>
in 1.4.37 and at least 1 flavor of 1.8). So my suggestion is that you<br>
either devise some workaround for this or set up multiple queues so you can<br>
feed calls to these "phantom-busy" folks. My "Expertise" (such as it is) is<br>
at the AGI level; I only fool with the portions of the actual tree code that<br>
are patently obvious (usually tweaks to patches).<br>
<div><div></div><div class="h5"><br>
<br>
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</div></div></blockquote></div><br>