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Still same thing nothing different Learning is not enough, you have to apply it...

It's seem either port connection issue or permission issue. Check the port 143 if it is open or not lsof -i -P I am not sure but your port should be 443 instead 143? Learning is not enough, you have to apply it...

I even tried call forward with analog phone instead of phantom but still not working. I am seeing the calling person number instead of our phone system main number. Any guide ? Learning is not enough, you have to apply it...

yes I did but same thing :( Learning is not enough, you have to apply it...

I created CLID and associted with the phantom TN. the CLID is below : ESA_HLCL 1111111 ESA_INHN YES ESA_APDN NO HLCL 1111111 DIDN YES DIDN_LEN 6 HLOC LSC CLASS_FMT DN Still no luck Learning is not enough, you have to apply it...

Hello, I am planning to accomplish almost same thing like this thread thread798-741566 Our company number is : 1111111111 Person A : 2222222222 DID : 1234567890 Extenal number : 1472583690 Person A dial DID 123456789 ---> IDC --> it's goes to phantom TN --> external number When the external...

check your voicemail.conf [General] pollmailboxes=yes pollfreq=10 Learning is not enough, you have to apply it...

I see that you are trying to get answer...but I have never used Cisco phone with Asterisk Sorry...hope somebody out there who might help. Learning is not enough, you have to apply it...

Thank's a lot Mr. Trouteaud. One more star from me. It will be really helpful as we are planning to migrate to CS1K 7.6 Learning is not enough, you have to apply it...

If your virtual TN is 100.0.0.14 Can you print this tn either going on ld 20 or ld 11 depend on your system ld 20 prt tn 100.0.0.14 Your type should be Let us know. Learning is not enough, you have to apply it...

Check this out : http://sourceforge.net/projects/jphonelite/ I have use a lot. It has not too good interface but have all the best feautres you might need and most of all, it is open-source. Good luck Learning is not enough, you have to apply it...

Wow... it was a qucik answer Just for that I am giving you star :) That's what I am worried about because right now we are using the SL1. So that's mean : Learning is not enough, you have to apply it...

Hello, We are planning to change the PRI provider for our phone system. We are going to use DB15 from Bell. As I learnt from this thread thread798-1579431, I will check IFC of each DChannel to make sure it match the protocol of the provider Anyone have an idea which Dchannel protocol does...

BTW, is there any Avaya conference unit which support IP (I mean without SIP) Finally, if you can provide me the step/information to configure the SIP phone in CS1K in case I need it, it will be awesome Thank's Learning is not enough, you have to apply it...

Is it worth buying licence for 1 sip conference unit Learning is not enough, you have to apply it...

I know but the thing is that most of our office, they do not have anymore analog phone connected. Learning is not enough, you have to apply it...

Do you any suggestion for any conference unit which use IP Thank's Learning is not enough, you have to apply it...

We are looking to add Polycom Soundstation Duo on our CS1K system. I would like to know : how I can setup it. I want to use it as IP Phone with PBX not with SIP (we do not have licence) Also, if you have any recommendation for a IP Conference unit, please feel free to share, Thank's for any...

Once you out the name, you have to put , to separate the name and family name Hope that help's Ques. Answer REQ new TYPE NAME CUST 0 DIG <enter> (empty) DN XXXX (dn ex: 1212) NAME FIRST NAME, LAST NAME Learning is not enough, you have to apply it...

So that's mean that the company setup on the elevator to dial the extension. I have to contact them to know how they setup it... And a FTR HOT D is it a good idea to put on the elevator. Thank you Learning is not enough, you have to apply it...