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About russelln

  • Birthday 04/10/1956

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  1. Thank you very much for the link, that looks simular to the page that was on the wiki, except it had a some more examples I think.
  2. I am setting up a demo system that I carry around and it is interfaced with an intercom and contol system that uses sip phones, just need to be able to demonstrate the delivery of voice mail to the email account with out having to use the WAN. thanks for the names of the products.
  3. thanks again, that is why I have been looking for a reference manual, all this info should be in book somewhere, I dont mind buying the book, as long as it is within reason $30-$50. I used to go to the pbx wiki and look at the examples there. thanks again,
  4. Hi, I hope this is the correct place for this question, I am looking for a simple email, cleint/server that I can install on a local computer on my Lan that can receive the emails from the CS410/425 and then user on the lan can log in get their emails, or have them forwarded to their own email address. Any ideas on a simple one to use? Thanks
  5. thank you very much, did not know case of the letter could be a problem, so I will try "1xx" to "sip:\*1xx@\r;user=phone;line=2" the reason I was hoping to use a buzy or off hook signal to hang up the line, is this unit I am connecting to does not send a CPC signal to connections I am trying to use, but it does send an off hook signal to get someone's attention to hang up the receiver. This unit is expecting both parties to hang up their recievers and send the hook switch back to it. If the CS unit would reconize the off hook busy tolence, it would know to dissconnect the line. thanks russell
  6. Ok, I am going to answer my own question here to show everyone I am not a lazy bum, ,I found the answer about 40 or so posts back, relating to a 411 call, still looking for the dial plan reference manual if there is such a thing. the syntax for the replacement is "105" goes to "sip:\*105@\r;user=phone" so for all extensions I want to call , so now how can I use "1XX" goes to "sip:\*1??@\r;user=phone" so I do not have put in 30 of these? also looking for still how to direct the call to a paticular pstn port on the CS 410 so my 2/3 digit numbers go to the lines they respond to and my real telephone calls go to the telephone co. thanks for any direction.
  7. I have been looking around the web site and cannot for the life of me find the dial plan information that was on the wiki,,i need to send the "*" to an outside port on a CS 410/425 to contol another deivce, my idea was to dial a 2 or 3 digit number and have it replaced with "*" to preceed it. you dial "105" and the switch sends "*105", I would like to be able to send them to a certian pstn port on the CS410 so any matching 3 digit number would be routed to say port 4 only. i need to know what the escape code for sending this out the port would be and where are the example locations now? is it also possible for the cs410 to disconnet on a busy cadence? i am using, bytheway the intercom works on this version. thanks
  8. "It seems like if you upgraded to 3.4, the extensions that were there before can't intercom to other extensions...something like that." Well really, the intercom went away with 3.2, cause that is where I have stayed. Permissions, multiple registrations, samehere, I know that, phones just ring... , from what I understand you have to buy software upgrade protection if you want to use the newer software, so that to me is having to pay to play..I would just like to have the intercom working again... I am thniking about pushing the system back to 3.1, I don't believe I rember I was having much trouble with it, I think I still have a copy somewhere. I guess I will do a backup and try it.
  9. hi,, I have read a lot of posts about the intecom featue not working after upgrading to 3.2, I am running, and it quit working on my system, I upgraded a while back and found that I have to pay to play, so I pushed my system back and never found out if the intecom feature worked with the new version, thing is I feel I paid for this feature when I bought the equiment, I feel it should work, one ,my system has been stable for almost a year now and two, I really do not want to spend a couple of my days of my life fighting with new problems from another upgrade. I am running 7.1.35 on my snom 300's and they all seem to behave properly and I have an 820 on the system with it's latest and greatest firmware and it does exactly what it is told to do. I am sure by now the problems have been solved, so what is the solution to the problem, I would like to be able to receive what I have paid for without spending my life tied to this little black box. I feel upgrades should not create problems but solve them...I am tired of having to holler into the other office to get a simple question answered when the IC sould be left on for that. I was wanting to sell and install these things, but I live in a part of the world that where people expect things to work as advertised when they are paid for and are not as tolerant as I am. thank you to whoever responds...
  10. Well out here in Scrub brush and tumbleweeds of West Texas, when it ocassionally rains, or if it the wind quits blowing, the constant assualt on the phone system causes static on the POTS lines... It seems that the little CS410 is interperting this as an incomming call...the analouge phone does not ring, it knows the difference between the 90v ring cadence and static (if the static is bad enough it will ocassionaly make a ding noise)...but alas the little box does not and I get phantom calls which never hang up. This may not really be the problem, but this weekend it rained and it drove me nuts Saturday with static calls. This causes the ports to hang and the only way to terminate the call is to break the connection from the box. Also if you are not there to answer the call then you get lots of messages over the weekend with static on them and both ports are hung up..It seems like in the past there was a version of the software that had a test for an active line in the port setup...now all we have is the CPC pulse and line impedence, it would be nice if there was some way for the port to really know if a real ring voltage was present on the port, instead of just a change of voltage. Usually about a couple of times per week I get the no service message and have to unplug both lines to get the ports to hang up. This has been going on for a while now, I just had some time to whine...
  11. Hi, I upgrade to the new version and now have 3 minute phone calls?? I pushed my system back to the old software and was ok for a day and today again I have 3 minute phone calls??? any clues as to what is going on? Russell
  12. is it possible to get a different message depending on from the extension on the state of the account? if you are on the phone and you get a call, can the call hear a different message than when the phone is just not answered? sernario 1, the incomming call gets the voice mail because the extension is busy and they hear "I am currently on the phone, please leave a message and I will call you right back" sernario 2, in incomming call gets ringing and then gets the voice mail and they hear " I am currently away from my desk, please leave a message and I will call you right back" is this possible with PBXnSip? Thanks
  13. So the software will not clear the lamp after listening to messages automatically? It seems werid that just deleting one message will do it, the server knows the messages have been listened to, because they become saved messages, it seems like the server would send the lamp off control message after listening to your messages. thanks, Russell
  14. I have tried to set up my phones, Snom, all 300 series, to have the message light blinking when their is a new message, and after the messages are listened to, to have it go off. What I have found is that if you do not delete at least one message after receiving a new message the light will not go out. Any ideas?
  15. Yes, I can make the calls when dialing from the phone.
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