Hi, There is no RBT in the incoming call from ISP to CUCM. There are no 180 Ringing or 183 Session Progress in call leg (ISP - CUBE) but in call leg (CUBE-CUCM) it is present. Could you please help me to resolve this issue? Scheme: ISP (83.137.x....
Hi, There is no RBT in the incoming call from ISP to CUCM. There are no 180 Ringing or 183 Session Progress in call leg (ISP - CUBE) but in call leg (CUBE-CUCM) it is present. Could you please help me to resolve this issue? Scheme: ISP (83.137.x....
Extension 0399 is in a partition of All Phones and the Calling Party Transformation CSS is set to Transform the extension to 0400 under Caller ID for Calls from this Phone. Our voicemail pilot is 7000. When calling voicemail with the transformation...
Dear Friends, I have problem for recording of SIP 3rd-party Phone with Cisco Mediasense because as I know , Mediasense only support in BIB mode(not SPAN mode) that SIP 3rd-party phones are not compatible. Could you please advice? Regards.
Hi Experts, We have a CME 2911 Router and around 45 Ip phones has configured. recently there is some of internal issues occurred. There is any way to Record all voice call into FTP Server or anywhere Details : Router : Cisco IOS Software, C2900 ...
Good morning, I currently have a problem with a phone, that runs on cucm 11.0.X, which does not show the active load in the cucm device configuration. I found this problem, because the phone has some trouble to upgrade to a new firmware. But that...
I have a 2800 with two PRI's using a separate carrier for each. Clocking is setup as: PRI connections on: 0/0/0 and 0/3/1 Clock Config: Network-clock-select 1 t1 0/0/0 network-clock-participate wic 0 network-clock-participate wic 3 slip errors ...
Hi All,Please i am unbale to dialout the call from my inrastructure.when dialing out we get the below mention CCM cause code. CCM SIP 503 SERVICE_UNAVAILABLE destination cause code and call get disconnect. call is not going to gateway.regards,Neeraj
Hi Guys, My Customer have the solution with CUCM 10.5.2 and 06 sites running SIP protocol. The Customer changed the protocol, previouly were H.323 and now it running SIP with change range Phone number. In any sites the Caller ID that is appearing i...
I have two sip trunks from different service provider, Line peer expires(sec) registered P-Associ-URI ================================ ========== ============ ========== ============ 154001 ...
I have testing on a Develop network for Cisco meeting server single split deployment. I want to use SipEdge feature on Edge Server. Is there anyone have the experience about SipEdge configuration? I have read the deployment doc for single split, but ...
Hi Guys , IS that possible to add support of CP-8800-A-Kem module to CUCM 8.6 ?
Dear All, I am planning to change cluster security mode from non-secure-mode to mixed-mode to enable call encryption, but before making any changes is there an option for me to test this on a few phones. Testing this as part of proof of concept ...
Hi, I am trying to make a e164 translation and translate +91.xxxxxxxxxx numbers to 0xxxxxxxxxx. So I am making a RP with /+91.xxxxxxxxx and removing predot and adding 0 as a prefix and then sending the call to the H232 gateway. But when I am diall...
Hi team! after long time struggling with dropped sip-ua registration authentication, our voice ISP advise to create "straight" trunk without auth...but with "binding over our public IP" actually, i don't now how to to do it...where and what needs to ...
We are running CUCM 9.1.2 on a publisher and one subscriber. When searching for CDR records in CAR we can only find records for the 3 phones registered to the publisher. The CDR enable flag is definitely enabled on both publisher and subscriber. We h...
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