CME 8.6 with IP phone and Analog Phone (Cisco 3825 + Vic2fxs )
Good day experts,
I am not an expert and would like to get help with my objectives.
What I have is IP phone and Analog phone connected to 3825 (CME 8.6) and connected to SIP ITSP.
What I want to do is for IP phone and Analog phone to ring at the same time and who ever answer first gets the call. The rests will stop ringing.
I have heard this is called Parallel Hunt in CME 8.6 but I would like verification for this.
I successfully setup to only redirect incoming call to FXs port but not to IP phone at the same time by using dialpeer. I hope this is possible since I have already invested to the router, fxs card and dsp card. I have a feeling that the easiest setup would be to get ATA box for analog phones.
Also can some one please send me a copy of your's config file for CME 8.6 that successfully configured that is similar to my objective? Like Dial peers, 911, and other stuff specifically for Toronto area if possible? I really appreciate if somebody can help me cause i have been pulling my hair lately. Thanks.
Yes, you can configure a blast/parallel hunt group. This is what the configuration would look like for a hunt group -
n the following parallel hunt group example, when callers dial extension 1000, extension 1001, 1002, and so on ring simultaneously. The first extension to answer is connected. If none of the extensions answers, the call is forwarded to extension 2000, which is the number for the voice-mail service.
voice hunt-group 4 parallel pilot 1000 list 1001, 1002, 1003, 1004 final 2000 timeout 20
You can tweak the above sample as per your requirements. Let me know if you got any questions related to this.
Hello all I have successfully converted analog phone into sccp. It has now its own extension number of 300. But now I need to be able to dial outside. How can I do this without assigning any of the phone with my full DID number with cme 8.6. I believe its called masking or translating my 3 digit extension into my 10 digit DID number so that my ITSP will allow me to route my call. Can somebody please help?
SIP traces provide key information in troubleshooting SIP Trunks, SIP
endpoints and other SIP related issues. Even though these traces are in
clear text, these texts can be gibberish unless you understand fully
what they mean. This document attempts to br...
Please find the attached HTML document, download and open it on your PC.
This provides an easy to use form where you simply answer a few
questions and it will render the proper jabber-config.xml file for you
to copy/paste. There is built in logic to verif...
CUCM Database Replication is an area in which Cisco customers and
partners have asked for more in-depth training in being able to properly
assess a replication problem and potentially resolve an issue without
involving TAC. This document discusses the bas...