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- About wmelick
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12-12-2018
18
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05-23-2019
01:24 PM
This worked. We use user device profiles (UDP) for PCCE. I did a UDP remove/add to pguser like three times and Finesse always said device not accessible. After removing the UDP from pguser, then un-checking the Allow CTI control in the UDP config, then adding back to pguser and re-checking the Allow CTI control, Finesse now logs the agent in with no issues! Amazing how a little process detail like this fixes an issue.
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Created by
wmelick
in Contact Center
in Contact Center
05-16-2019
01:15 PM
05-16-2019
01:15 PM
Thanks for the explanation!
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Created by
wmelick
in Contact Center
in Contact Center
05-09-2019
12:30 PM
05-09-2019
12:30 PM
I have just started doing some SQL queries to our PCCE AWDB using MS SQL Server Management Studio. I love the access to historical data on specific calls and the ability to do joins on other tables to create very useful reports. One thing I noticed when tracing a RouterCallKey in the Termination_Call_Detail table is that there is a lot of TalkTime reported when the AgentPeripheralNumber is NULL. I am assuming that this means a caller is inside a script and is hearing media (hold music or announcements) during this time? Later in the flow (on a different RouterCallKeySequenceNumber) I see the call eventually get routed to an Agent....so it looks like this intermediate TalkTime is when they are in queue. I see the database schema guide defines TalkTime as "The cumulative time, in seconds, that the call was in a talking state on the destination device. TalkTime is a completed call time, not an agent state time. " I am wondering if the system considers media playing in a script as "talking state" when not connected to an agent. Can anyone confirm what this TalkTime data means? Thanks.
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02-28-2019
10:51 AM
I think this is a fairly basic question, but I am trying to send outbound fax calls from an FXS port to a PRI on a Cisco router (configured as H.323).
I have the inbound config OK (assign the FXS voice port an internal station ID, setup an inbound translation rule changing the DID to the station ID, setup a dial peer destination pattern matching the station ID and set it to the FXS port).
How do I do the inverse to get the FXS port to route out the PRI using an H.323 config?
Thanks
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Created by
wmelick
in Contact Center
in Contact Center
02-13-2019
06:29 AM
02-13-2019
06:29 AM
Your response led me to an excellent doc I wanted to share with the community:
https://www.cisco.com/c/en/us/support/docs/customer-collaboration/unified-contact-center-express-1151/211376-Technote-on-How-Bosh-Connection-Works-fo.html
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Created by
wmelick
in Contact Center
in Contact Center
02-13-2019
06:15 AM
02-13-2019
06:15 AM
Thank you Jonathan. I will check that out....
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Created by
wmelick
in Contact Center
in Contact Center
02-12-2019
11:24 AM
02-12-2019
11:24 AM
Does anyone know the mechanism Finesse uses internally to check the reachability of an agent? I am asking because we are getting some reports of agents on VPN (AnyConnect) getting disconnected from Finesse. Is there something like a timer which every X milliseconds checks the agent client app running on the browser?
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Created by
wmelick
in Other Collaboration Subjects
in Other Collaboration Subjects
02-12-2019
11:19 AM
02-12-2019
11:19 AM
I couldn't quite get this working even with TAC support, but the closest I got was "HIGH:!EXP:!MD5:!RC4:!ADH:!EXP:!LOW:!3DES:!MD5:!aNULL:!eNULL:!AES128-SHA:!AES128-SHA256:!AES128-GCM-SHA256:!AES256-SHA256:!AES256-GCM-SHA384:@STRENGTH" which our audit tool only listed as moderate vulnerabilities (no severe or critical). The TLS-RSA ciphers were still being negotiated. However since there is no CVE associated with these cipher suites, our Info Sec team was OK living with the moderate vulnerability.
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Created by
wmelick
in Other Collaboration Subjects
in Other Collaboration Subjects
01-09-2019
05:58 AM
01-09-2019
05:58 AM
My Info Sec team wants to disable some specific TLS-RSA ciphers on our expressways which do not support forward secrecy. I am not very familiar with the cipher suite syntax but was wondering if I can append a "-" and the specific cipher suites I want to disable? For example, if I append ":-TLS-RSA-AES128-CBC-SHA:-TLS-RSA-AES128-CBC-SHA256" to the cipher suite, will that then disable these two specific ciphers? I just want to make sure I have the correct syntax if this method will work.
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Created by
wmelick
in IP Telephony and Phones
in IP Telephony and Phones
12-20-2018
01:48 PM
12-20-2018
01:48 PM
We didn't notice it initially with Jabber either because Jabber's default settings are to enable video for calls (options->calls->Always start calls with video). This setting is compatible with Verizon cell phones. So people test dial with Jabber and say "it works fine."
If that Jabber option is set to "Never start calls with video" then the audio fails. Adding the "voice-class sip audio forced" to a matching dial peer enables audio on either Jabber video setting.
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Created by
wmelick
in IP Telephony and Phones
in IP Telephony and Phones
12-20-2018
01:41 PM
12-20-2018
01:41 PM
Adding "voice-class sip audio forced" on the CUBE dial peer seems to have resolved all our issues with 8845/8865 phones and Jabber clients (11.9 and 12.5) dialing Verizon cell phones.
The Jabber issue was not initially noticed because Jabber's default settings (options->calls ->Always start calls with video) are compatible with Verizon cell phone audio. It's only if that setting is modified to "Never start calls with video" that the audio issue is noticed. (basically mimicking the 8845/65 camera shutter open/closed condition...open = audio, closed = no audio).
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Created by
wmelick
in IP Telephony and Phones
in IP Telephony and Phones
12-17-2018
08:48 AM
12-17-2018
08:48 AM
Thanks for all the replies. We have discovered that Jabber phones (tested on versions 11.9 and 12.5) also have the same issue calling Verizon cell phones....but we have not been able to find a workaround in any of the Jabber client settings. Jabber calls just do not complete to Verizon cell phones. And yes, our ITSP is Verizon. One of my co-workers has a TAC case open....I will update if they provide any recommendations which fixes the issue.
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Created by
wmelick
in IP Telephony and Phones
in IP Telephony and Phones
12-13-2018
06:39 AM
12-13-2018
06:39 AM
After testing, we discovered that if the 8845/8865 camera shutter is closed when a call is initiated, no audio will pass to Verizon cell phones (only affecting Verizon). Calls to Sprint and T-Mobile cell phones work normally with no issues. However if the shutter is open when the call is initially placed, or if the shutter is opened mid-call, then audio will start flowing normally to Verizon cell phones. If the shutter is then closed mid-call, audio will continue to flow normally through the call. Given the circumstances, I suspect there must be some video capability message (no video available when the shutter is closed) being sent in the SIP packets which Verizon interprets as no audio capability (or to mute audio). Other carriers are not affected by this video capability message or ignore it or something. Again this happens only with firmware sip8845_65.12-1-1SR1-4 but not with the older firmware sip8845_65.11-5-1-18 . 8841 phones do not have video and are not affected.
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Created by
wmelick
in IP Telephony and Phones
in IP Telephony and Phones
12-13-2018
05:29 AM
12-13-2018
05:29 AM
Thank you --- I will look into that and post an update if I find a resolution.
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Created by
wmelick
in IP Telephony and Phones
in IP Telephony and Phones
12-12-2018
02:01 PM
12-12-2018
02:01 PM
Just wondering if anyone else has seen this issue: 1) We just upgraded 8841/8845/8865 phone firmware from sip8845_65.11-5-1-18 to sip8845_65.12-1-1SR1-4 on our cluster 2) After the upgrade, outbound calls from 8845's and 8865's to Verizon cell phones have no audio in or out. Calls to other carriers like Sprint and T-Mobile work just fine. Calls to landline phones work just fine. 3) Calls are going through a SIP cube. We have run some SIP traces and see no codec problems or dead streams comparing successful audio calls to no audio calls. 4) Calls from 8841 phones running the newer sip8845_65.12-1-1SR1-4 work just fine --- to the same Verizon phones and others 5) If I downgrade the 8845/8865's to the previous firmware sip8845_65.11-5-1-18 then calls to the Verizon cell phones work fine again with two way audio
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05-23-2019
This worked. We use user device profiles (UDP) for PCCE. I did a UDP
remove/add to pguser like three...
05-09-2019
I have just started doing some SQL queries to our PCCE AWDB using MS SQL
Server Management Studio. I...
Created by
wmelick
in Unified Communications Infrastructure
05-03-2019
05-03-2019
Just FYI this tool still works well with Windows 10 and Unity 11.5.1
Created by
wmelick
in Unified Communications Infrastructure
03-28-2019
03-28-2019
I just wanted to add that I was getting the ODBC connection error as
well. I was using my own Unity ...
02-28-2019
I think this is a fairly basic question, but I am trying to send
outbound fax calls from an FXS port...
02-13-2019
Your response led me to an excellent doc I wanted to share with the
community:
https://www.cisco.com...
02-12-2019
Does anyone know the mechanism Finesse uses internally to check the
reachability of an agent? I am a...
02-12-2019
I couldn't quite get this working even with TAC support, but the closest
I got was
"HIGH:!EXP:!MD5:!...
01-09-2019
My Info Sec team wants to disable some specific TLS-RSA ciphers on our
expressways which do not supp...
01-08-2019
My Info Sec team wants to remove some specific TLS-RSA ciphers which do
not support forward secrecy....
Created by
wmelick
in IP Telephony and Phones
12-20-2018
12-20-2018
We didn't notice it initially with Jabber either because Jabber's
default settings are to enable vid...
Created by
wmelick
in IP Telephony and Phones
12-20-2018
12-20-2018
Adding "voice-class sip audio forced" on the CUBE dial peer seems to
have resolved all our issues wi...
Created by
wmelick
in IP Telephony and Phones
12-17-2018
12-17-2018
Thanks for all the replies. We have discovered that Jabber phones
(tested on versions 11.9 and 12.5)...
Public Statistics
| Date Registered | 12-12-2018 01:33 PM |
| Date Last Visited | 08-16-2019 07:14 AM |
| Total Messages Posted | 18 |
Helpful Votes Given To
| User | Helpful Count |
|---|---|
| 5 | |
| 5 | |
| 5 | |
| 10 | |
| 5 |
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