Bill H

Any thoughts on the above situation with CO lines and DID?

The new CS-410 does not have the DID setting. What would be the new method? I would like CO1 and CO2 to go to Hunt Group 111 and CO3 and CO4 to go to Hunt Group 222. Thanks...

OK I figured it out. On the CS-410 If I added the port I wanted to use on the end of the SMTP Server setting it works. Ex. smtp.comcast.net:587 However, I did try this in the version prior to 3.1.2.3120 (Linux) and I believe that it did not work. As many of us know, when testing and observing, the results if unfavorable are sometimes forgotten. Bill H

How can I change the SMTP Port? I looked at the PBXNSIP GUI and did not see a setting to change. Bill H

If you are using Windows. Try turning the Firewall off in the PBXNSIP computer. Bill H

I am a complete know nothing on Linux. Which version of Linux does the CS-410 use? There are 4 different versions on the PBXNSIP website available. Thanks

I creted a new extension and used the Static Registration method to direct calls to the cell. This works OK. I put this new extension in a Hunt Group (Stage 1) and set the Duration for 5 seconds. I added an existing extension in Stage 2. The cellphone will ring when I dial the new extension number it is registered to but it continues to ring after 5 seconds. The Stage 2 extension never rings at all, even after the Stage 1 Duration is well expired. Any Clues??? PS I also did the same test using an Agent Group and had the same results. Also, are we still limited to PSTN for a destination? Will SIP Trunks ever be addressable? Bill H

I have worked with the Snom M3 and somewhere along the line I have seen the Aastra unit. The GUI for each product is strangely similar, if not identical. This leads me to believe that they may be the same units with different firmware. In my opinion, I would buy more M3 base units to spread the load around before choosing Aastra. The M3 is much less expensive and may produce the same results in terms of radio range. Bill H

In Trunk A, set the Failover Behavior to "On all error codes" I believe that will work with your current Dial Plan. Bill H

OK I now see where and how to get what I need. However, when an extension logs out and then back in again, they are placed at the end of the Agents list. We use the "Use preference from the Agents setting" to place our best agents up front. Loging in and out of a specific ACD will place them in last place. Any idea on future releases to put them in their pre-logout position? Bill H

Yes that is correct. EIther method will do fine.

I'm not sure if I posted this before. Now that an extension can log in and out of individual ACD Goups I was wondering where I could look, programmatically, to see what an extensions staus is in a particulat ACD Group. In the Database I can see if the extension is logged in or out, but that is Global from *64 and *65 Feature Codes. Bill H

In your Dialplan create a SINGLE DIGIT CODE for the trunk you want use for Forking. Example: Pref____Trunk____Pattern___Replacement 999___Forking______7*_________* Next: On your Accounts / Redirection page enter 7 (in this case) and then your cellphone number in the "Cell Phone Number" area. Forked calls will now use the selected trunk. Bill H

Let me start off by saying that you have an improper connection. Ideally you should use a FXS Adapter like the Grandstream HT-502. (It has 2 FXS ports, supports t.38 and is inexpensive) Although your method will allow you to send and receive faxes, there are trade offs such as you are experiencing. Check and make sure the fax is dialling with DTMF tones and not Dial Pulse (DP). Dial pulses produce a "Kickback" (collapsing field) voltage that may be interpreted by the CS-410 as an incomming ringing signal. Sometimes even simply picking up and hanging up on the shared line can trigger the same results. Also, when the fax is in use (in or outbound) you presently have no indication on the phones (LED's). And since there is no Privacy with your configuration and you happen to pick up while the fax is in use, you will hear the fax data and may even disconnect it. Another possibility is that IF the CS-410 has any type of "Fax Detection" ability it will hear the tones and begin to ring the phones. You said: "why does the box answer when sending out going faxes, it has never done this before, and I have not changed a thing?" Well I'm sure that you didn't change anything but your local telephone service provider MAY have. Something as simple as replacing or transferring your service in a cable miles away from your location can change the PSTN line characteristics. Maybe even an "Upgrade" made to some outside equipment could be the cause. My suggestion is to use the proper method and get some type of FXS adapter and you will see the trouble disappear. Bill H

There are two ways to do this: You don't use the Caller ID Name. 1. When the caller presses 1 have the AA send the caller to a ACD Group that has a Display Name "Gold Account" When the caller presses 2 have the AA send the caller to a different ACD Group that has a Display Name "Silver Account" etc.... Set the "From: Header" in the ACD to "Group Name (Calling Party)" When the caller is answered from the ACD Group the "Group Name with Calling Party" will appear. 2. You can use the same method with "Hunt Groups" if you don't want to have them wait in a Que Group. I have not tried this out myself, but we do use a similar method. So I'm not sure if "Calling Party" is the Caller ID Name or Number. BillH

The name of the person you are calling is sent back to your phone in the Session Descrition Message where it says "To:" Here you can see where I (From:Bill H) called 204. SIP/2.0 183 Ringing Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK529adcf0b4a33dd7f.6dce497b54b3b09a0 From: "Bill H" <sip:205@xxx.xxx.xxx.xxx:5060>;tag=23bf47af02 To: "204" <sip:204@xxx.xxx.xxx.xxx:5060>;tag=7450add047 Call-ID: a97b2bcc6cd8a1a9 CSeq: 14255 INVITE Contact: <sip:205@xxx.xxx.xxx.xxx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2998 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 277 v=0 o=- 61201 61201 IN IP4 xxx.xxx.xxx.xxx s=- c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 60172 RTP/AVP 0 8 18 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv That being said I believe that PBXNSIP is sending the Name to the telephone (in the SDP), but the telephone lacks the ability to extract and display it. Aastra has a common Directory configuration file that all our phones pick up by TFTP when they boot up. Does Polycom have such a method as well? Bill H

Sorry I forgot to include that. We are using the Web Link method. Bill H

Click to Dial doesn not work. Switched back to previous version to get it working. Bill H

I see some interesting additions here. Are the Release Notes available? Bill H

"i know that there is an option to offer camp on but will this notify the caller that the person is on the phone?" Yes. It will say that the person is on the phone and you (the caller) can press 1 for a call back (Camp On) when the person is done with their call or press 2 to leave a message in the Voice Mail Box. _______________________________________________________________________________ "or will it give the same message whether they are on the phone or just no picking up?" If they are not on the phone then there is NO Call Back Offer Message, it will ring until the Call Forward No Answer Timer expires and the call goes to the Voice Mail Box. Bill H

I too have been working on a program to do the same. I have seen something like this too. But I don't remember seeing "Waiting". I have seen "Connected" as a State. What I get from the ACD SOAP is the Message when the caller enters the ACD. It shows a 0 (Zero) time. There is also information about the call id and the ACD Group they enetered into. Then when they are leave the ACD I see the time they were waiting. It never tells me which extension the call went to or if they hung up. This raw data when a call comes into an ACD Group: POST / HTTP/1.1 Host: xxx.xxx.xxx.xxx Content-Length: 418 SOAPAction: IvrInput Content-Type: text/xml Accept-Language: en-us Connection: Keep-Alive Keep-Alive: 5 User-Agent: Mozilla/4.0 (compatible; PBX) <env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://soap.com/pbx"><env:Body><sns:ACDWelcome><Queue>sip:562@localhost</Queue><CallID>4b74e2ed20dabf0d#af739ed194</CallID><From>"Bill Hayhurst" <sip:205@localhost></From><To><sip:562@localhost></To><Duration>0</Duration><State></State><Extension>205@localhost</Extension></sns:ACDWelcome></env:Body></env:Envelope> Any help???? Bill H

We have the same situation here. I believe that it has to do with the registration timer in the phone. We get a series of ? (question marks) where we would normally see a phone icon. I think the phone carries some internal information that it saves concerning the dialogs. When PBXNSIP is shut down and re-started there is new dialog information that the phone does not have internally. If I remember correctly, it will resolve itself over a period of time. You could also set the Auto Resynch to midnight (phone will Auto Re-Boot) and that would most likely solve the problem. I think that if the registration timer in the phone DOES NOT expire before PBXNSIP goes back on line may be part of the problem. Look at the dialog registration time and determin its maximum. As a atest, try and keep PBXNSIP down for about that much time plus a minute if possible. This will allow the registration timer in the phones to expire. Let us know what happens. Bill H

What type of phones are you using? We had the same situation and resolved it by using the Directory feature in our Aastra telephones. At that time, about a year ago, PNXNSIP did not provide the Called Party info to the receptionist. Maybe it has changed since then. So, look in your phone and see if it has any type of local directory that can be programmed with your extension numbers. Bill H

The reason you can't get multiple calls now is because once your Extension is busy it is not considered AVAILABLE by the ACD Group. Therefore, no multiple calls to your extension from the same ACD Group. Placing it on Hold doesn't change its status as far as the ACD Group is concerned. The easiest thing to do is to give him 3 different extensions on his phone, if that is possible. Then assign the 3 extensions to each of the 3 ACD Groups in tha same order. Set the Number of agents added per stage: to 1 in each of the 3 ACD Groups. Or, you can use Hunt Groups. They ring through like gangbusters even if the extension is busy. But hey have any Announcement ability. The ACD has a little class going for it and you can have multiple announcements. Bill H