tag:blogger.com,1999:blog-2158853543793456735.post4195376377066303058..comments2023-10-02T04:50:01.667-07:00Comments on VoIPNorm's Collaboration Blog: Updated: CUCM SIP URI Dialing to Lync 2013–New SIP URI Normalization rules on CUCMChris Normanhttp://www.blogger.com/profile/07200178774058910421noreply@blogger.comBlogger24125tag:blogger.com,1999:blog-2158853543793456735.post-40292875194841666292015-01-21T06:22:32.304-08:002015-01-21T06:22:32.304-08:00Hi Chris,
A quick question. If the user hits igno...Hi Chris,<br /><br />A quick question. If the user hits ignore in Lync when a call comes from CUCM via his RDP Lync sends back a SIP DECLINE message and CUCM drops the call for good. Normally the expected behavior is if a remote destination rejects the call it should still ring the main Cisco phone until NOAN timer expires. Any comments on this and how can we fix it? <br /><br />Thanks a lot!<br />Alexandru Z.https://www.blogger.com/profile/06450737181871611955noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-9595266619784279142014-08-11T23:03:53.477-07:002014-08-11T23:03:53.477-07:00Great article!
Hey Tony did you sort this out i a...Great article!<br /><br />Hey Tony did you sort this out i also have the same problem with 9.1.2. Marknoreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-447380623769393262014-07-19T14:41:12.804-07:002014-07-19T14:41:12.804-07:00Hello, in a semi-related question, has anybody got...Hello, in a semi-related question, has anybody gotten Twilio SIP to work for inbound directly to CUCM without CUBE? I got it working outbound perfectly (for use by lync 2013) thanksCryptosmasherhttps://www.blogger.com/profile/17671081688967668352noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-1563845167647119852014-06-25T18:53:28.200-07:002014-06-25T18:53:28.200-07:00Hello Chris,
I agree with everyone here that your...Hello Chris,<br /><br />I agree with everyone here that your blogs are fantastic. Thank you for sharing your knowledge. We went ahead and followed this guide and setup SIP URI dialing. It is working beautifully. I noticed something about the call flow and was wondering if this is expected or not. So, the scenario is that we have two Lync users in the data center, User1 and User2. User1 dials User2's extension in his Lync client and this in turn rings User2's Cisco phone and User2's Lync client. If User2 answers the Lync client, the the snooper trace shows that the ICE negotiation is happening with the Mediation Server's IP address. I was assuming the ICE candidate IP address would be that of User1. So, the media flow is happening through the Mediation Server in this scenario. Is that expected behavior when using SIP URI dialing method and RDP ? Thanks in advance.Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-44101017834214314282014-03-19T03:12:33.693-07:002014-03-19T03:12:33.693-07:00It is really not easy to do this kind of configura...It is really not easy to do this kind of configurations but mostly companies have experts who do only this kind of work, I think they done best what they can. As I am a user of IP phone <a href="http://www.internetvoipphone.co.uk/grandstream-dp715-dect-handset-and-base-station.html" rel="nofollow">Grandstream DP715</a> when ever I have an issue the company who is providing me services they come to me and resolve issue. Hope they will come back to you.Anonymoushttps://www.blogger.com/profile/10864714321870179068noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-13971756277324733732013-12-25T13:59:14.649-08:002013-12-25T13:59:14.649-08:00Can you please help to transform Request URI from:...Can you please help to transform Request URI from:<br /><br />INVITE sip:6101;phone-context=cdp.udp@umc.ua:5060;maddr=172.20.245.196;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0<br /><br />to that Request Uri:<br /><br />INVITE sip:6101@172.20.245.196:5060 SIP/2.0Sandyhttps://www.blogger.com/profile/00213193774399005778noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-39351705233123121692013-09-25T07:38:27.657-07:002013-09-25T07:38:27.657-07:00Thanks for such a great integration article - I ha...Thanks for such a great integration article - I have this working (mostly) with Lync 2013 and CUCM 9.1.2. There is a delay for some users when calling their Cisco extension and having RDP ring their Lync phone. We are seeing a few "SIP/2.0 403 Forbidden" with a reason of "Application accepts invitations via static registration only." using ocslogger / snooper. I can't find anything useful through Google, wondering if you have ever see this error / delay or have any suggestions?<br /><br />Thanks!Anonymoushttps://www.blogger.com/profile/14120729834805164804noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-73553922003027312012013-08-02T05:51:47.711-07:002013-08-02T05:51:47.711-07:00We need a consultant who can help us to make this ...We need a consultant who can help us to make this configuration work. Please, send a mail to dklimenkov@temenos.com.<br /><br />Best regards,<br /><br />Dmitry KlimenkovAnonymoushttps://www.blogger.com/profile/08112636210031472465noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-7280130469351511082013-07-16T03:21:08.745-07:002013-07-16T03:21:08.745-07:00very helpful artical
Generate Your revenue with ...very helpful artical <br /><br /><a href="http://voipsuggestion.blogspot.com" rel="nofollow">Generate Your revenue with iTel| Boost Your revenue with iTel Product |</a>Voip Suggestiionhttp://voipsuggestion.blogspot.com/noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-16921376590621628732013-07-11T07:52:12.162-07:002013-07-11T07:52:12.162-07:00Hello Chris,
Thank you very much for the great st...Hello Chris,<br /><br />Thank you very much for the great step-by-step documents. I am now working on integrating LYNC2013 and CUCM 8.6.2. The outgoing calls from LYNC are working perfectly well, but the incoming i cannot make to ring simultaneously on the Cisco phone and LYNC. I configured the second trunk exactly as you've proposed but it does not work. It rather rings ONLY on the Cisco phone with enabled remote destination, but if the Cisco phone is off, the call fails. I tried to trace the call to SIPUri trunk but could not even find the trunk name in the tracelogs on call mananger. Can you, please, advice me where to look for the further investigations as i love the idea of SIP URI remote destination. DO not want to deal with a separate extensions pool for dual-forking.<br /><br />Thank you very much in advance,<br /><br />Best regards,<br /><br />Dmitry. (klimed@gmail.com)Anonymoushttps://www.blogger.com/profile/08112636210031472465noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-85385040810026102472013-07-02T23:04:56.309-07:002013-07-02T23:04:56.309-07:00Hi,
Great blogs! They have been very helpful to me...Hi,<br />Great blogs! They have been very helpful to me as Ii am in the middle of a Lync 2013 - Cicso integration project (about 1000 users with a pilot of 250 lync plus cals).<br />Currently our Cisco partner is only pushing the interop capabilities of RCC with our cisco environment, which to me totally defeats the purpose of the advantages that lync brings over jabber/webex.<br />I think I will trial the 3 integration methods that I have found – cucilync, rcc and dual call controls (with sip uri dialling! )<br />What sort of click to call control is possible between lync and the cisco handsets with a dual call control configuration?<br />On a side note, your fantastic blog has inspired me to write one about a win I just had with publishing lync via citrix xenapp. I have gotten the vdi plug in to work while delivering a rich ICA experience from our xenapp servers (despite a distinct lack of good blog articles about it).<br /><br />Cheers<br />Matt<br />Anonymoushttps://www.blogger.com/profile/07025003686076650749noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-52613874581817717062013-07-02T17:11:46.969-07:002013-07-02T17:11:46.969-07:00Hello Dave,
I have tried to send u n email. But a...Hello Dave,<br /><br />I have tried to send u n email. But address seem to be wrong. Can you please take contact with me on evyn.valayten@eis.mu. I really need your expert advise with lync 2013.<br /><br />Cheers,<br />EvynEvyn Valaytenhttps://www.blogger.com/profile/01752676199234802889noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-64157033547402047522013-07-01T00:48:08.272-07:002013-07-01T00:48:08.272-07:00Hi Chris ,
We've been planning to configure s...Hi Chris ,<br /><br />We've been planning to configure setup between Lync 2013 and CUCM. please send me some screenshots related setup. maxcoder1 at gmail dot com Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-33667638931597399162013-06-28T08:44:13.795-07:002013-06-28T08:44:13.795-07:00Thanks Simon. Looks like a great addition to the s...Thanks Simon. Looks like a great addition to the solution. I will post it up as an update when I get a chance to collect some more info on how to implement.Chris Normanhttps://www.blogger.com/profile/07200178774058910421noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-34696431373899566442013-06-27T13:57:50.652-07:002013-06-27T13:57:50.652-07:00Chris, Anonymous, We were facing the same issue wi...Chris, Anonymous, We were facing the same issue with the Invite coming in as user@domain.com:5060 from CUCM. When you stated this was fixed by setting the port to 5061 I presumed you had a mediation pool so could use 5061. We don't have mediation pools with each our pools so we looked on the Cisco side and came up with the following normalization script to associate with the trunk to change default 5060 to 5061.<br /><br />M = {}<br />function M.outbound_INVITE(msg)<br />local method, ruri, ver = msg:getRequestLine()<br />local uri = string.gsub(ruri, "5060", "5061") <br />msg:setRequestUri(uri)<br /><br />end<br />return M<br />Anonymoushttps://www.blogger.com/profile/18176267513645614640noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-15719111560672936772013-06-27T13:55:20.888-07:002013-06-27T13:55:20.888-07:00This comment has been removed by the author.Anonymoushttps://www.blogger.com/profile/18176267513645614640noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-53557223304446986122013-06-17T10:03:03.675-07:002013-06-17T10:03:03.675-07:00Hello again Chris,
Thanks for your help (previous ...Hello again Chris,<br />Thanks for your help (previous writeup on the SIP-URI dialing). Lab setup was working great, however a Cisco person told us there is a limited number of RDP available... this lead us to try RCC (since they are installing CUPS). it worked nice, other than losing the ability to use the Lync soft phone when dialing out. Then I saw this update with the addition of the ms-skip-rnl in the lineURI. When i tried to add this entry to an existing user, it returned an error.<br />I don't have DID numbers assigned, I also don't have extensions assigned. Will this command work without extensions?Mike Whttps://www.blogger.com/profile/13876291339821775765noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-11487331023566532502013-05-02T14:14:13.469-07:002013-05-02T14:14:13.469-07:00Hi Michael,
I am investigating your issue a littl...Hi Michael,<br /><br />I am investigating your issue a little further. Wondering if you could email me how you exactly addressed this issue. chris.norman@hotmail.com<br /><br />Let me know how you setup port 5061 on the mediation server side. I am assuming that because you has a separate mediation server that you could set the listening port to 5061 but I want to make sure. If you can send me some screenshots of your setup that would be great.Chris Normanhttps://www.blogger.com/profile/07200178774058910421noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-87077664531264881102013-04-01T11:26:11.445-07:002013-04-01T11:26:11.445-07:00Talking about SIP-URI dialing feature, can I dial ...Talking about SIP-URI dialing feature, can I dial any desired SIP-URI on world with it without preconfiguring each domain in dial plan?Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-12113537006497462042013-02-25T08:24:54.832-08:002013-02-25T08:24:54.832-08:00Thanks for the update. Its a great way to configur...Thanks for the update. Its a great way to configure the environments when you have an existing legacy Cisco install and layering on UC with Lync.Chris Normanhttps://www.blogger.com/profile/07200178774058910421noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-65152604255829392512013-02-25T06:38:18.869-08:002013-02-25T06:38:18.869-08:00UPDATE:
I raised a case with Premier support and t...UPDATE:<br />I raised a case with Premier support and the engineer has lead me to add UserServices in the SIP Trace. This helped us find this error:<br />“Exit: Port already present; and it is not 5061; reject request”<br /><br />After changing the SIP Request Uri to port 5061 on Cisco side, the issue was resolved and calls now succeed in all cases - does not matter in what pool the user is.<br /><br />I conclude that this configutation works perfectly and provides exactly the capability that we needed. You get the best of both Lync and Cisco.Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-23185866749865563082013-02-23T12:29:57.097-08:002013-02-23T12:29:57.097-08:00Hi Chris
In the case when user is in another pool...Hi Chris<br /><br />In the case when user is in another pool, The front-end server rejects the call to Mediation server via response containing:<br />Start-Line: FailureResponseException: ResponseCode=504 ResponseText=Server time-out<br />DiagnosticInformation=ErrorCode=2,Source=FE-servername,Reason=See response code and reason phrase<br /><br />The "Server time out" is also logged as the reason for the failed call in monitoring reports.<br /><br />I have separated mediation server pool, no co-location. I also have mediation server pool as part of the other FE pool, but it is not integrated to this same CUCM that I talk about.<br /><br />This brings me to idea that potentially - if you needed to host users in the other pool, (and Lync turns out not able to handle this rerouting natively), you can add use different "virtual" SIP domain in the RDP profile for the user to route these calls to the different trunk - and remap the domain back to normal Lync SIP domain via Lua script just before sending to Lync. Would work but not very clean.<br />Well ideal would be if this routing happens natively in Lync. <br /><br />Michael<br /><br />Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-10806668793757616132013-02-23T10:49:54.345-08:002013-02-23T10:49:54.345-08:00Hi Michael,
How does Lync reject the call back to...Hi Michael,<br /><br />How does Lync reject the call back to CUCM? Does it return a SIP message back to CUCM to say it dropped the call?<br /><br />Also is this a collocated mediation server on the FE? and do you have a mediation server setup on the other pool?<br /><br />BTW I have not heard or seen this before, but its good info to know because multiple pool scenarios is something I had not tested and may require a little different design.<br /> <br />Chris Normanhttps://www.blogger.com/profile/07200178774058910421noreply@blogger.comtag:blogger.com,1999:blog-2158853543793456735.post-40271493088882588512013-02-23T06:47:19.638-08:002013-02-23T06:47:19.638-08:00Hi Chris
I am very happy to see such article, as ...Hi Chris<br /><br />I am very happy to see such article, as this is exactly what I am trying to setup. Thank you and well done writing up these great articles! I have overcome some challanges already (multiple trunks to centralized CUCM with media bypass in branch sites, and splitting the SIP-uri for same domain to multiple trunks) but one issue is still oustanding. I am curious if you have come accross this. If you use sip-uri dialing the mediation server trace indicates this:<br /><br />-MEDIATIONSERVER<br />MediationCall: 4d6a991bf1724f7ca99d08d2ebdf0a23<br />CallId: 3c5afe00-1281bdd5-76c5b-11e6fc17@x.x.x.x<br />From: sip:+44444444@x.x.x.x<br />To: sip:firstname.lastname@contoso.com:5060<br />Direction: Inbound<br />Start-Line: Invalid phone number for inbound call, pass it along.<br />$$END-MEDIATIONSERVER<br /><br /><br />The call is still passed to the front-end pool and succeeds. <br /><br />However, if the user is in another pool, the call is still delivered to the local pool and not the users's pool. The local pool FE seems to log this and drop the call:<br /><br />LogType: diagnostic<br />Severity: warning<br />Text: Message was discarded by the application<br />Result-Code: 0xc3ee7964 ES_E_REQUESTURI_VALIDATION_FAILED<br />SIP-Start-Line: INVITE sip:firstname.lastname@contoso.com:5060 SIP/2.0<br />SIP-Call-ID: a5c03a60-e790-4490-9e02-75baf13b8250<br />SIP-CSeq: 51567 INVITE<br />Data: application="http://www.microsoft.com/LCS/UserServices"<br />$$end_record<br /><br />BTW it's Lync 2010.<br />Do you know if any special format of the request Uri is required? Like adding ;user=phone, or ;phone-context=enterpise, or something like this?<br /><br /><br />Michael<br /><br />Anonymousnoreply@blogger.com