Hello All I built a sip trunk to a telephone provider's PBX. They ask me sent " invite" include both G711ulaw and G711alaw. I checked" Media Termination Point Required" . There only is G711ulaw in "invite" message. Do you have any experience abou...
Hello All I built a sip trunk to a telephone provider's PBX. They ask me sent " invite" include both G711ulaw and G711alaw. I checked" Media Termination Point Required" . There only is G711ulaw in "invite" message. Do you have any experience abou...
Cisco 9951 phone is not registering with cme 9.1. When I run debug ccsip messages, I am getting below errors Received:REGISTER sip:10.20.1.1 SIP/2.0Via: SIP/2.0/UDP 10.20.1.95:5060;branch=z9hG4bK2bc83d5cFrom: <sip:4500@10.20.1.1>;tag=64e9502f6135003a...
Hi all, I've been configured Phone Template with SURL on the 2nd button.When I'm adding new phone from template it's taking the SURL just like It's supposed to be,but when I try to add another phone from template there Isn't SURL, It's gone from the ...
Hopefully someone can answer this. I am on CUCM 11.5 and CUCA... I have a directory handler set up for our main call handler. My CEO would like it if when someone spells her name in the directory and chooses it, that it goes to her AA's extension i...
Hello masters! First of all, thank you for your help.I am not very familiarized with VoIP implementations or configurations, so I would like to know how can I transform or redirect an incoming call from a number of the PSTN to an internal directory n...
Long time reader, first time poster. Be gentle. I am running into an issue with inbound dial-peer matching in our multi-tenant config on our CSR1000v. I originally started with matching by host IP address, but if i have 2 separate customers that are ...
Hello together We are currently changing the Telephony of one of our customers from analog to VoIP.We have already used the ata 191 several times and are very satisfied so far. Now the customer has a doorbell, which is connected to the analog System....
Outbound calls coming into a CME which are blind or consult transferred over a SIP trunk to a CUCM show the number of the CME extension initiating the transfer instead of the original calling number. If the call is blind or consult transferred to an...
I has a centralized UC infrastructure with only two CUCM and remote sites with Cisco 4321 voicewayand some 7821 IP phones. The PSTN is linked to the gateway with an E1 on the local sites and the gateway uses H.323 to the CUCM. There is one DID to...
Hi, my Cisco Jabber disconnects randomly throughout the day. I get "Connection to Phone Service failed. Error Code: CJ:100:2I called my help desk and they stated it was my internet. However, I have no other issues with any other software. I am on zoo...
Curious if anyone has built or setup a solution that allowed you to bridge multiple FXS ports in one simultaneous call? We have a paging system with a unique campus setup where each of the units is a master for that building but does not connect to ...
Hi All, Just before I go fighting my way through TAC, can I presume the ISO for V14 on CCO is not bootable still and is NOT suitable for an upgrade from V11? Thanks All
Using a daisy chain kit between two 8832 with a physical separation betweens tables in the room. Looking for information on USB-C cable lengths allowed between Smart Adaptor and 8832 units. Are there any extender options for these connections? The Sm...
Hello,Is there an easy way to broadcast a voicemail message to all mailboxes within Cisco Unity?
Hi I was wondering if it is possible to Select "Forward All" and then select a speed dial number and have the extension forward to the corresponding Speed Dial Key? I tested and it works on a regular number or extension. The issue I am having, is tha...
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