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