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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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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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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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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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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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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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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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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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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