Resolved! Call Express 7915
Can you tell me how to change and update a BLF-Speed Dial on a 7915 Call Express? I found the config and shows as ephone 26 , template 1.Thank you
Can you tell me how to change and update a BLF-Speed Dial on a 7915 Call Express? I found the config and shows as ephone 26 , template 1.Thank you
Hi,I'm wonding if TCL script is supported for incoming calls on SIP line. If not what is best alternate to play greeting messages.Anees..
Hi,Just want to know that like we change the codec of Cisco or non Cisco IP Phones in cme environment by giving command " codec g711alaw". But in CUCM can we change the codec per phone or not ? Thanks
Dear Experts;I am very new to asterisk.We have a contact center and customer needs to record all the calls from or to the agents.I have created a SIP trunk and recording profile as per the call manager 7.1 documenation and Installed Free Asterisk PBX...
Hi All,We are trying to register two sip trunks with two different providers. We managed to get them registered, but with the configuration in the cisco document, both credentials are sending to both providers, which results in a provider blocking ou...
Hi,i have problem .In srst mode all the calls get connected successfuly except the international numbers and the reason is the the last digit of the number is getting stripped which is very strange !dial-peer voice 70 pots description SRST Internatio...
Hi Experts,In my LAB, gateway is configured as mgcp in CUCM with one E1 pri.I want to configure one FXS port for FAX machine in this gateway.what configuration is required on mgcp GW and at cucm end.Plz help.sajeel
Hi Allwe have created the following xml file and copied all images to flash but cannot change the background image the phones.<CiscoIPPhoneImageList><ImageItem Image="TFTP:Desktops/640x480x24/tharawat640.png" URL="TFTP:Desktops/640x480x24/tharawat640...
My senarios is we have a CUBE (2921 router 15.1) which establishes two SIP trunks with two SIP providers. And it registers with CUCM 9.1 as H.323 gateway. Voice mail is Unity Connection 9. all incoming calls are translated to the hunt pilot number (n...
I have an existing Callmanager system connected to an Asterisk box that has been running for quite some time.Just recently a problem came up that I'm having difficulty tracing down.Problem is when SOME IP phones (mostly 7940's) call an inside extensi...
please gudie me how can we enable voice packet compression in Asterisk/Voipswitch with G729/G723 codecs baiscally i want to compress voice packets in to 8/10kbps how can i do this please guide me awaiting thanks.
I am very new to IP VoIP, and I am starting to have an issue with my 7911 and 7961 phones not booting up. These are phones that were active and just stopped working. When I try to power them up using either POE or a power adapter, they will flash som...
Hello, I have a 2801 SIP gateway which connects to my ITSP.SIP authentication etc seems to be fine. I can call out, but I cannot receive calls.Here is the output of "show sip-ua register status":Line peer expires(sec...
%SIP-3-BADPAIR: Unexpected event 21 (SIPSPI_EV_CC_CODEC_LOCAL_DNLD_DONE) in state 25 (SIP_STATE_MIDCALL_RECD_SUCCESS) substate 0 (SUBSTATE_NONE)We are getting dead on transfer through CVP. I know it has something to do with SIP stack. and something t...
Hi, Once a week, since the installation of the unity, I got a major issue. The Unity 7.0.2 Crashing. When that event happen the only way to put unity back running is to restart the server.Im unified with Domino DUC 1.2.4 and the Lotus notes versio...
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| Subject | Author | Posted |
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