I opened a case with Cisco for this and they provided workaround which works for me:   Workaround: Local SPAN is not available on the ASR, but creative use of ERSPAN can produce similar results. An example configuration is given below:int loopback0 int loopback0 ip address x.x.x.x y.y.y.y ! ! monitor session 1 type erspan-source source interface GigabitEthernet0/0/0 destination erspan-id 100 ip address x.x.x.x origin ip address x.x.x.x ! ! monitor session 2 type erspan-destination destination interface GigabitEthernet0/0/5 source erspan-id 100 ip address x.x.x.x   This is as per defect https://bst.cloudapps.cisco.com/bugsearch/bug/CSCto80065   ... View more

For quick and cost effective implementation of Chat with Finesse it might be worth checking CrowConnect solution: https://www.crowconnect.com/site/docs Regards, Leszek ... View more

I know this has been asked a long time ago but maybe someone will find this useful. If you have the topology like: Phone----CUCM----CUBE----ITSP then the following setting works just fine and achieving what is expected by telco: On CUCM: On route pattern: Calling Line ID PresentationRequired Field to  Restricted Calling Name PresentationRequired Field to Resticted   On SIP trunk check: Remote-Party-Id and asserted-Type change to PAI   This way CUCM will send to CUBE: P-Asserted-Identity: "Leszek Wojnarski" <sip:+888888888@10.10.10.10> Privacy: id   And then on CUBE need to put on dial-peer to telco something like: voice-class sip asserted-id pai voice-class sip privacy-policy passthru   this is just to forward all the information related to PAI as it to ITSP.     Leszek ... View more

Configuration seems to be fine from what I could see and you CUBE should ate least attempt to Register to TELCO. The only thing that still can be checked it if your CUBE can ping/resolve both registrar servers: 7002619856.siptrunking1.appiaservices.com you might want to try to define this via IP if that's possible in your case: registrar ipv4:10.10.10.10 expires 180 So basically your CUBE should start sending REGISTER messages just after  "registrar" command is added under sip-ua. So the action plan I'd suggest: 1. Try to ping your servers via DNS. 2. If you can ping then remove registrar commands from sip-ua. 3. Enabled debugs ccsip messages. 4. Add registrar commands at this point your CUBE should start sending REGISTER messages. It will try couple of times and the stops trying if this fails. Leszek ... View more

Hi Ben, From debugs I don't see anything wrong with the Authentication setup itself, it looks more like issue with the format of the number or something like this as telco is sending: 480 and 404, which would mean that they don;t have this destination number available. So I'd suggest checking with telco to verify what number to the expect (how many digits and what prefix). Or thye might be able to tell right away once you show they the debugs from this of why they send 480 and 404, as from our perspective we only receive it and cannot really tell why, they should tell. Jan 27 15:33:16 EST: //2438/DDA892000000/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 480 Temporarily Unavailable Call-ID: AA655189-E40611E6-8285DF6E-B03DE5EA@207.32.250.245 CSeq: 102 INVITE From: "Collaboration Room" <sip:574XXXXXXX@207.32.250.245>;tag=143CB468-1459 To: <sip:1574XXXXXXX@205.X.X.X>;tag=sip+1+c8c80055+2ea3b9ff Via: SIP/2.0/UDP 207.32.250.245:5060;received=207.32.250.245;branch=z9hG4bK12421D2 Content-Length: 0 Supported: resource-priority,siprec Contact: <sip:205.X.X.X:5060> Server: DC-SIP/2.0 Organization: Metaswitch Networks Jan 27 15:33:16 EST: //2439/DDA892000000/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.X.X.X:5060;branch=z9hG4bK1251D21 From: "Collaboration Room" <sip:574XXXXXXX@172.X.X.X>;tag=143CB6E8-1A9E To: <sip:1574XXXXXXX@10.X.X.X1>;tag=203~c2551668-bbbd-4dfe-a185-df4baa6ae3ae-47421692 Date: Fri, 27 Jan 2017 20:33:34 GMT Call-ID: AAC6F838-E40611E6-8287DF6E-B03DE5EA@172.X.X.X CSeq: 101 INVITE Allow-Events: presence Reason: Q.850;cause=1 Server: Cisco-CUCM11.5 Session-ID: 00000000000000000000000000000000;remote=738dce7f7600a1cae75d8bac7cac6ba0 Content-Length: 0 HTH, Leszek ... View more

Hi Ben As per my understand of what's expected from telco (I don;t know what's been agreed with them), is the call flow similar to: http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-aaa.html Figure 3. UAC-to-UAS Call Flow with INVITE Message In your case CUBE doesn't send second INVITE or we cannot see it. Can you send the debugs: debug voice ccapi inout and debuc ccsip messages from the time when you try to make a call. You mentioned that you have two types of failures that are happening so it would be good to have debugs from both. Leszek ... View more

I'd probably go with MediaSense + Calabrio, you can buy bundle directly from with Cisco and they can give you fairly good discount. And with plans to have Screen Recording with MediaSense 12 it would be complete solution in my opinion. Or you can just do MediaSense only and buy interface from some other company, but that that depands on the features you need. Leszek ... View more

Can you sen the output from: sh sip-ua register status And additionally try to configure: authentication username 7002619856 password 7 xxxxxxxxxxxxxxxx on the dial-peer level? Leszek ... View more

Hi Ben, Can you share your GW configuration (full if possible), but specifically: sip-ua, and dial-peer sessions would be nice to see. Leszek ... View more

Ritesh, Can you provide some more details so that everyone can understand the issue better: - did this ever work? - you are using SPAN based recording on BIB recording? - you've mentioned you take pcap, where do you take it, on Verint itself?  - I don't know much about Verint but usually those systems would require CTI integration to get CTI events about particular calls, is the CTI user created and all the phones are added. Leszek ... View more

First try to isolate the issue, in your case it can be on the GW or CUCM. Run debug on the GW: debug h225 asn1 debug h245 asn1 debug voice ccapi inout Once we have those then we should be able to tell if the issue is on CUCM or GW, once we know that then we can do more focused troubleshooting. Leszek ... View more

What codecs do you use for point to point calls? Is VERINTConfigured to accept whatever codec is needed. If you use 722 you might try to disable it in enterprise parameters or on the phone itself, and use 711 instead. Leszek ... View more