Bill H

When I have a message in my mailbox is it possible to turn on the Message Waiting Light on an additional telephone. This way the secretary would know when I have a message.

This is using the internal PSTN gateway of the CS-410. Should there be any SIP Messaging going on here between the gateway and the CS-410???

CS-410 3.2.0.3143 (Linux) When and incomming call is transfered to a cell phone the caller hears an announcement "This number could not be found" after a minute or so. Followed by "Please enter the destination number followed by the pound key". The Xfer procedure is: Answer incomming call Press Xfer Dial cellphone number Speak to person on cellphone Hang up The caller is then transfered OK and they cal talk to the cellphone until the announcement comes on and the cellphone is disconnected. Is there a different procedure for this type of transfer???

From the Wiki: <env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://soap.com/pbx"> <env:Body> <sns:ACDWelcome> <Queue>sip:73@pbx.company.com</Queue> <CallID>9479cafa-18966f55-cd0021a8@192.168.1.113#e510828f0c</CallID> <From>"Fourty Eight" <sip:48@pbx.company.com></From> <To><sip:73@pbx.company.com></To> <Duration>16</Duration> <State>connected</State> <Extension>48@pbx.company.com</Extension> <Agent>42</Agent> </sns:ACDWelcome> </env:Body> </env:Envelope> We are using 3.3.0.3152 (Win32) and I am not seeing <Agent>XX</Agent> in my SOAP message.

Here is what I have in mind until this is potentialy added as an option. The ringing the caller hears while being transfered comes from the ring.wav (I believe) I can mix music into it there. The bad part (if considered bad) is that extension to extension calls will also hear the music during the ring cycle. Do you see any additional pitfalls seen here?

I have searched through PBXNSIP looking for this but I can't find it. When a call is transfered from one extension to another, the caller hears a ringing sound. Is it possible for the caller to hear the Music On Hold sound instead? Thanks

Thank you for explaining that feature.

Yes that part I did figure out. But what is the meaning/function of this particular item? "When the allocation of a new CO-line failed and a call was rejected because of this:"

Could someone please explain this new fdeature: From..... Email Event Notifications: "When the allocation of a new CO-line failed and a call was rejected because of this:" Thanks...

Regular Expressions have always interested me. Yours is simple and clean. Could you explain each element in detail. Thanks....

I see the problem here. The CS-410 is using its local LAN Address (192.168.1.99) in the SDP and that is an unroutable address from the remote stations standpoint. I have resolved this problem with other products using STUN. The customer has only 1 Public IP Address. This is from the WIKI: If you don’t have a second IP address things are getting a little bit trickier, but you can still deal with this. You can add an additional IP routing entry in the PBX service (not in the operating system) that will tell the PBX that routing to a specific IP address will result in a specific IP address. This is done in the setting "IP Routing List" which is just below the "SIP IP Replacement List". Is this something I should be seeing in the CS-410??? --------------------------------------------------------------------- So it sounds like I would plug the LAN of the PBXNSIP into the router and give it STATIC 192.168.1.99 / 255.255.255.0 / 192.168.1.1 for IP Assignments and also plug the WAN of PBXNSIP into the same router and give it STATIC Public IP Address / Public IP Addresses SNMask / Public IP Addresses Router asignments. Is this correct?

I am connecting to a CS-410 as a remote station. I can see the remote stations Remote IP Address (Ex. 71.211.22.33) on the Account Registration page At the remote station I can call a cellphone and they can hear me, but I can't hear them. When I looked at the SIP Trace the Session Discription Protocol shows the remote device with the Inernal IP address of the PBXNSIP not the external IP addreess I found. How do I correct it be the correct External IP Address?? The CS_410 is connect to the customers LAN on its LAN Port.

Pending Plan: We have an application where tech support is provided over the phone to customers. Each customer has an account number which reflects the amount of support time they are entitled to. If their time expires during a support call the call should be cut off unconditionally. How, if at all, can we implement this into PBXNSIP? I thought of a Calling Card, but the WIKI is a little light on it. Any ideas???

This is from the latest PBXNSIP Know How email: Concerning Que Groups Call Priority - 'Know How' - This can be the Queue Entry time OR if required, try and pull out specific callers when the agent who last answered the call becomes available. I don't seem to be able to understand what is being said here.

Yes I have those settings set. I re-tested it just to be sure. Click on dial does not add the leading digit 1.

How do I add Agent Queue Groups to be monitored by the PAC? I was able to add extensions, but see no area to add Queues.

That did not resolve the trouble. I can manually add a 1 to the telephone number for a customer in Netsuite by editing the account. Before: (212) 235-1234 After: 1 (212) 235-1234 Then the Click To Dial Works Ok only when using the older version. The only problem here is that we have over 30,000 customers, faxes, cellphone numbers to edit!!!!! I installed the latest version you made available 3147. It appears as if it still isn't adding a leading digit 1 to the number in Netsuite. However using the 3147 version, when I clicked on 1 (212) 235-1234, in Netsuite, it looks like 3147 removed the leading digit 1 instead of adding it. Hope this helps. PS I tried different setings in the trunk used to call out on. Check Country Code/ 10/11 digit dialling.

I discovered this too on the CS-410 with the 3144 version. Dial *90# and it will ask you to "Please enter the extension number" Then it will work like an intercom. Why has this been changed?

We upgraded to V 3144 and now our Click To Dial is not working. We have tested and determined that the leading digit 1 is not being applied/applied. We use Netsuite for our CRM. It has a click to dial feature which works well with PBXNSIP. The telephone numbers in Netsuite are only 10 digit numbers (212) 235-1234. In our older version of PBXNSIP it added the needed leading digit (1) and the calls went through. I tried each setting in the Trunk/ "Rewrite global numbers" with no success. Any ideas????

I am almost certain that I have seen the answer to this question, but I just can't find it on the forum. I would like to send the Simple CDR to 2 different IP Addresses. I tried a SPACE a COMMA a SEMICOLON between the two addresses and I could not get it to work. Is this possible? If so, what is the delimiter between the 2.

I think I am missing something. Is this the same as the Simple CDR? I mean the area where you establish the CDR String you want by selecting the $ characters? Example: $c' inserts the caller-ID of the remote party. $d' inserts the duration of the call in seconds. $s' inserts the duration of the call connected to an extension. If so, then what is the $Character for the filename of the recording and does it also give you the Index of the recording? Thanks

Here is an old telecom trick. Instead of dialling *67 dial 1167 it may work for you. 1167 is the Dial Pulse equivelent to DTMF *67. It was used for those that did not have Touch Tone (DTMF) dials long long long ago. Bill H

OK I will try out the different domain for a "second" business at the same location. Can the "Seek" option be made selectable?

On the subject of expanding the feature: MOH on CS-410 When the first caller is placed and remains on Hold they hear the MOH Music/Announcement from the beginning. During that time additional callers placed on Hold will drop into the "Loop" and will start to hear what ever is playing at that instant. Virtually all telephone systems use the "Loop" method for Music/Announcements On Hold but PBXNSIP is unique by starting from the beginning. Well, kind of. This is OK, but businesses today want to fully exploit this captive audience area (Caller On Hold) and deliver a structured and precise message. A caller placed in a "Loop" announcement may get the tail end of the message and miss the real impact of the main greeting. Now, is it possible to have each caller that is placed on Hold receive the MOH from the beginning? This would certainly set PBXNSIP apart from the crowd and be virtually impossible for traditional TDM systems to duplicate. Lets push it even further: How about a different MOH file for each CO Line? Customers often times have a "second" business running in the same office. This would allow each CO Line to play the correct MOH message to callers. There is such an add-on item available, but it is unreliable at best. Your thoughts???

Thank you. That worked well. I didn't know where the DIDs were. Bill H