Ha! Thanks Vip! Sorry about not including my version numbers too. On my production box I'm using 1.8.3 (that's the debug from the original email). On my demo box I just build I'm using 1.8 SVN-trunk-r309404 and that's what generated these logs. I'm not sure if this is a chan_sip.c problem or if this is a dial plan problem.
So digging in a bit deeper, Asterisk is receving the real REFER message. The "REFER-TO: <sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2787?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto-tag%3D8be38bb187>" is accurate and in chan_sip.c it knows how to manipulate it. It does grab the "from-tag" and "to-tag" and parses the data. On one of the lines below you can see it says "Looking for Call ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 (Checking From) --From tag 15826bef52 --To-tag as41bacc0b". Then it moves on to bridging the peers/channels together. It's not until later that I get the final " SIP/2.0 481 Call leg/transaction does not exist" which doesn't make sense to me. Also, the Lync client says "Call was not transferred because [Original Extension] cannot be reached and may be offline." <-------------> [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 0 [ 53]: REFER sip:1820@10.10.10.10:5060;transport=TCP SIP/2.0 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 1 [ 78]: FROM: <sip:1...@lyncserver.internal.name:5068>;epid=E5790B0758;tag=15826bef52 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 2 [ 41]: TO: <sip:1820@10.10.10.10>;tag=as41bacc0b [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 3 [ 13]: CSEQ: 2 REFER [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 4 [ 58]: CALL-ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 6 [ 59]: VIA: SIP/2.0/TCP 20.20.20.20:5068;branch=z9hG4bK70e8a145 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 7 [107]: CONTACT: <sip:lyncserver.internal.name:5068;transport=Tcp;maddr=20.20.20.20;ms-opaque=09aa43d8a2a895b9> [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 8 [ 17]: CONTENT-LENGTH: 0 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 9 [200]: REFER-TO: <sip:lyncserver.internal.name:5068;transport=Tcp;maddr=20.20.20.20;ms-opaque=09aa43d8a2a895b9?REPLACES=a9b5f241-5e9d-4439-b347-2cac9384a627%3Bfrom-tag%3Daa19f11d4f%3Bto-tag%3D7a9abe27a5> [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 10 [ 40]: USER-AGENT: RTCC/4.0.0.0 MediationServer [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: --- (11 headers 0 lines) --- [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: = Looking for Call ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 (Checking From) --From tag 15826bef52 --To-tag as41bacc0b [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: Call 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 got a SIP call transfer from caller: (REFER)! [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Attended transfer: Will use Replace-Call-ID : a9b5f241-5e9d-4439-b347-2cac9384a627 F-tag: aa19f11d4f T-tag: 7a9abe27a5 [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: SIP transfer to extension lyncserver.internal.name:5068@from-internal-xfer by (null) [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: SIP attended transfer: Transferer channel SIP/Lync-00000003, transferee channel SIP/1820-00000002 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/1820-00000002' [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: <--- Transmitting (no NAT) to 20.20.20.20:5068 ---> SIP/2.0 202 Accepted Via: SIP/2.0/TCP 20.20.20.20:5068;branch=z9hG4bK70e8a145;received=20.20.20.20 From: <sip:1...@lyncserver.internal.name:5068>;epid=E5790B0758;tag=15826bef52 To: <sip:1820@10.10.10.10>;tag=as41bacc0b Call-ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 CSeq: 2 REFER Server: FPBX-2.8.1(1.8) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1820@10.10.10.10:5060;transport=TCP> Content-Length: 0 <------------> [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Trying to put 'SIP/2.0 202' onto TCP socket destined for 20.20.20.20:5068 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Looking for callid a9b5f241-5e9d-4439-b347-2cac9384a627 (fromtag aa19f11d4f totag 7a9abe27a5) [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Strict routing enforced for session 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: set_destination: Parsing <sip:lyncserver.internal.name:5068;transport=Tcp;maddr=20.20.20.20> for address/port to send to [Mar 4 12:54:53] DEBUG[11296] netsock2.c: Splitting 'lyncserver.internal.name:5068' gives... [Mar 4 12:54:53] DEBUG[11296] netsock2.c: ...host 'lyncserver.internal.name' and port '5068'. [Mar 4 12:54:53] DEBUG[11293] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/1820-00000002 Variable: SIPREFERRINGCONTEXT Value: from-internal Uniqueid: 1299261284.2 [Mar 4 12:54:53] DEBUG[11293] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/1820-00000002 Variable: SIPREFERREDBYHDR Value: Uniqueid: 1299261284.2 [Mar 4 12:54:53] DEBUG[11296] netsock2.c: Splitting '20.20.20.20' gives... [Mar 4 12:54:53] DEBUG[11296] netsock2.c: ...host '20.20.20.20' and port '(null)'. [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: set_destination: set destination to 20.20.20.20:5068 [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: Reliably Transmitting (no NAT) to 20.20.20.20:5068: NOTIFY sip:lyncserver.internal.name:5068;transport=Tcp;maddr=20.20.20.20 SIP/2.0 Via: SIP/2.0/TCP 10.10.10.10:5060;branch=z9hG4bK3f177f10 Max-Forwards: 70 From: <sip:1820@10.10.10.10>;tag=as41bacc0b To: <sip:1...@lyncserver.internal.name:5068>;epid=E5790B0758;tag=15826bef52 Contact: <sip:1820@10.10.10.10:5060;transport=TCP> Call-ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 CSeq: 103 NOTIFY User-Agent: FPBX-2.8.1(1.8) Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 49 SIP/2.0 481 Call leg/transaction does not exist -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Friday, March 04, 2011 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk <-> Lync / Call Center Transfer / Refer I feel your pain On Fri, Mar 4, 2011 at 9:29 AM, Danny Nicholas <da...@debsinc.com> wrote: -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro Sent: Friday, March 04, 2011 8:07 AM To: asterisk-users@lists.digium.com 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:us...@domainname.com <mailto:sip%3aus...@domainname.com> >" 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
