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Hi Experts, I have a question, in our env.(ISR 4431) there have no radius server(now we use syslog to send compact fields) but we have to identify which department the calls belong to. We'd like to mark department name in sip header but either in com...
Hi expertsmay I ask how to force rtp go through cube, our voip environment isgenesys — cube — cucm All established call is point to point rtp transmition, cube is only forwarding signaling , is there any ways to make rtp can through cube?
Hello Experts,
I have a question that if ISR can select dialpeer when detected some specific sip codes Such as 404 or 503.
I'm not saying about the SIP OPTIONS, i mean in some cases the remote side can properly respond options messages but for some c...
Hi experts,
We faced a problem that some user don't hang up the call so there will have some ports be always occupied...
Is there any possibility to limit call duration on ISR 4431 or 3945?
Like if call duration exceed 1 hour, ISR will initiatively d...
Hi Experts,
I have a question that could cisco isr automatically enable/disable dialpeer or translation-profile at a specific time period?
Or any other functions that can reject sip requests at specific time?
Thanks for reply, Jonathan. So you mean voice hunt would also work for sip environment? Previously I merely tried this command in ISDN-SIP environment, I never think of it can work under pure sip environment
Thanks, Haris. I know 'voice hunt' and in my understanding this command should use for ISDN calls. Can it be also used for sip calls? because I see the disconnect cause code should the ISDN cause code
Thanks for your reply, Dennis... But it's really hard if i want to trace a specific call, in my environment, VG 3945 connect 16 E1 and cps can almost reach 50 at peak hour...
