Bill H

Has anyone experienced a problem with the Voice Mail in the 2.1.0 version? If I call (*97#) into the VM and change my greeting I get a New Message as soon as I press # to save my greeting. The same action takes place when I change my Name in the VM. Again I receive a New Message. When I go to to listen to the New Message it is either empty or I hear the Prompts to record my name. If you have updated to 2.1.0, please try this and let me (and the Forum) know if you have the same experience. Thanks, Bill H

Has anyone experienced a problem with T1 service on inbound DID/DNIS/Alias calls with the new 2.1.0 version? Thanks, Bill H

My question: If I have 20 extensions in an ACD Group and I use 2/10 for the Call Rate Limitation in what order or method does the PBX distribute the calls? Does it select random extensions from the group? Does it review past activity on an extension to set its order? Thanks, Bill Hayhurst

I'm not sure if we are or not. I believe that Verizon can accept the whole string (*8212124567890) without a pause after the *82 input. If thats the case then it would look like single stage dialling. But lets say that Verizon needs a pause, after the *82 input, for about 1/2 of a second. Is that possible? Also, I am still not able to get the original Dial Plan to work. Today I will remove all other dial plans to be sure there is no conflicts. Bill H

Try dialling *82 and then your telephone number. I either get a "Call Failed" or a dial tone when finished dialling. Bill H

I have all the characters packed in tight. I don't see any spaces in the string. Bill H

The request goes to an external PSTN Gateway. I am trying to switch the Caller ID function from the Local Telephone Service Provider (PSTN) on a per call basis. Normally calls going out from our system are Caller ID Blocked at the PSTN Service Provider level. By sending *82 to the PSTN Provider (and then the dialled number), the Caller ID Funtion is turned on for that call to be identified. Bill H

Well I have been hacking away at this problem and finally went to the Logging area. [7] 2007/08/17 20:56:08: UDP: Opening socket on port 57004 [7] 2007/08/17 20:56:08: UDP: Opening socket on port 57005 [5] 2007/08/17 20:56:08: Identify trunk 3 [8] 2007/08/17 20:56:08: Resolve destination 253992: a udp 78.123.102.224 33665 [8] 2007/08/17 20:56:08: Resolve destination 253992: udp78.123.102.224 33665 [8] 2007/08/17 20:56:08: Send Packet 100 [6] 2007/08/17 20:56:08: Sending RTP to 78.123.102.224:3000 [8] 2007/08/17 20:56:08: Play audio_moh/noise.wav [7] 2007/08/17 20:56:08: UDP: Opening socket on port 49594 [7] 2007/08/17 20:56:08: UDP: Opening socket on port 49595 [8] 2007/08/17 20:56:08: Resolve destination 253993: url sip:*8212122261234@192.168.1.160;user=phone [8] 2007/08/17 20:56:08: Resolve destination 253993: udp 192.168.1.160 5060 [8] 2007/08/17 20:56:08: Send Packet INVITE [8] 2007/08/17 20:56:08: Resolve destination 253994: a udp 78.123.102.224 33665 [8] 2007/08/17 20:56:08: Resolve destination 253994: udp 78.123.102.224 33665 [8] 2007/08/17 20:56:08: Send Packet 183 [8] 2007/08/17 20:56:08: Resolve destination 253995: url sip:*8212122261234@192.168.1.160;user=phone [8] 2007/08/17 20:56:08: Resolve destination 253995: udp 192.168.1.160 5060 [8] 2007/08/17 20:56:08: Send Packet ACK [5] 2007/08/17 20:56:08: INVITE Response: Terminate 6a194e63@pbx [7] 2007/08/17 20:56:08: Other Ports: 1 [7] 2007/08/17 20:56:08: Call Port: f248812ade877381f1815ed26b5fa1b3@192.168.0.123#436a4874b8 [8] 2007/08/17 20:56:09: Resolve destination 253996: a udp 78.123.102.22433665 [8] 2007/08/17 20:56:09: Resolve destination 253996: udp 78.123.102.224 33665 [8] 2007/08/17 20:56:09: Send Packet 403 [8] 2007/08/17 20:56:10: Resolve destination 253997: a udp 192.168.1.102 5060 [8] 2007/08/17 20:56:10: Resolve destination 253997: udp 192.168.1.102 5060 [8] 2007/08/17 20:56:10: Send Packet 200 I can see that the *82 was entered ahead of the telephone number, but the PBXnSIP server sent me a 403 Forbidden code. Anyone know why this is happening and what to do to solve it??? I forgot to mention that this going to a external PSTN gateway. Bill H

I have been trying for a hour or three to get a particular Dial Plan to work for me. I just cannot seem to find the correct combination. The Object: To have *82 placed ahead of an 11 digit telephone number. The Pattern: Dial 9 and an 11 digit USA telephone number. The Replacement: Dial *82 and then the 11 digit telephone number. ------------------------------------------------------------ I tried 9* as the Pattern and sip:\*\1@\r;user=phone as the Replacement. The \* is supposed to use the * as a literal STAR in the dial plan. It just does not work. Will the PBX accept a literal at this point? I also used sip:77777777\1@\r;user=phone as a test Replacement and it did dial 77777777 and the 11 digit telephone number OK. Then I tried sip:\1\*82@\r;user=phone as a test Replacement and it did the 11 digit telephone number with the *82 appended to it. It looks like the literal can be added after the \1 and not before it. Am I missing something here??? Bill H

I am not 100% sure of what is taking place. SIP Trunks act differently than POTS and I know POTS better than SIP. Anyway....... It sounds like the "Number has been disconnected" announcement is coming from the ITSP and not PBXnSIP. You are "downstream' in the call flow sequence at this point while the ITSP has control over what happens when all 3 lines/channels are busy. From that position, I don't think there is much you can do. The fourth caller, most likely, never makes it to your location. What happens if you de-register the service and call the main number? Does each line Register as its own trunk or is there one registration for all three? Who is the ITSP? Bill H

I have been trying to get the Address Book to work also with no success. I started using the Address Book feature in the individual phones instead. It isn't as vast (200 numbers) as the PBXnSIP's Address Book. Depending on which manufacturer of phones you use, you can put the Address Book names and numbers in the <mac.cfg> for each phone or use the <all.cfg>. I use Aastra and there are 2 config files. The common one to all phones <aastra.cfg> (loads first) and the individual config files for each phone <mac address.cfg> (loads second). I did notice, at one time, that the Name and Number stored in the PBX Address Book would show up on the CDR when ever the number was dialled manually or the Speed Dial Number (*123 example) was used. That kind of indicates that it is in the PBX system doing something, but not attributing the Caller ID Number to a name listed in the Address Book. I hope this may help/ Bill H

I am working on an application that will receive digits from an IVR Output, take some action and then transfer the caller to an Auto attendant or another IVR. The SOAP connection is working by sending the request to my application but I keep getting the same Call Log info when I send back to it. It seems that the PBX can't resolve the Destination. 192.168.0.123 is the phone I am calling from. [8] 2007/08/02 20:53:28: Play recordings/ivr12.wav [6] 2007/08/02 20:53:28: Received DTMF 1 [6] 2007/08/02 20:53:29: Received DTMF 2 [6] 2007/08/02 20:53:29: Received DTMF 3 [6] 2007/08/02 20:53:29: Received DTMF # [8] 2007/08/02 20:53:29: Resolve destination 1046: url sip:101@192.168.0.123;transport=udp [8] 2007/08/02 20:53:29: Resolve destination 1046: a udp 192.168.0.123 [8] 2007/08/02 20:53:29: Resolve destination 1046: udp 192.168.0.123 5060 [3] 2007/08/02 20:53:29: IVR Node: Unknown call [8] 2007/08/02 20:53:29: Send Packet BYE I am using the latest revision of PBXnSIP on Windows. I followed the WIKI guidelines for sending the <Destination>444</Destination> and <CallID> & Callid$ & </CallID> (VB6 strings) and I can see it all of it going to PBXnSIP using EtheReal sniffer. Looking at the Call Log of PBXnSIP I see the above listing. Anyone else been to this point already??? Bill H

I have been using Callcentric (www.callcentric.com) for awhile and it works well. I had an intermittant "one way voice" trouble when I first signed on. My service wasn't even going through PBXnSIP at that time. They stuck with it until it was cleared up. Price??? You get what you pay for. I have 3 incomming channels (lines) for $8.95 / mo USD. Your needs may be different. You can also get a Callcentric Network number for free and use it in and out for free. If you have friends with SIP phones any place on earth you can talk for free

I don't know Polycom products that well, but you need to look for an Auto-Answer feature. 4.6.1.7.2 Ring type <ringType/> from Polycom manual. I use Aastra phones and they have it. Bill H

I would like to hear from any of you out there with PBXnSIP as well as any other IP PBX system installations. The questions I have are: Using an in-house network with no remote locations connected. How many of you require a seperate wire run to the IP Phone as opposed to "sharing" a wire with the computer? What condition would exist where this is needed? What advantage or disadvantages are there? Is it necessary to have the IP Phones and PBX on a completly seperate network, switches, wires and all? Or will it work Ok over an existing network? Lets say there are 25 computers and 25 telephones in this example and we are only concerned with the internal communications. Bill H

We have installed PBXNSIP in our office and everything seems to be working OK except that after awhile we see "No Service" display on all of the Aastra 57i phones. It seems like a loss of Re-Registration trouble. The PBX Minimum Registration Timer is set to the Default value of 30 seconds and Maximum of 360 (Default value). Each phone is set internally to 0 for the Registration Period. This allows the pbx to establish the registration time. Initially the phones come up OK, but after an undetermined amount of time they all stop working. I used the Ethereal sniffer and can see the initial registration and re-registrations take place, but have not yet caught the trouble. Is it possible that all 12 phones are trying to re-register at the same time (since the common timer is set by the pbx) and swamping the pbx? As a test, we have changed the PBX Minimum Registration Timer to 120 seconds and have staggered the Registration Period of the telephones by 1 second to minimize the rush of registrations. We are waiting to see if this resolves the trouble. Has anyone else seen a similar trouble, found a resolution or have any additional ideas of what items to test??? If any additional info is needed, please let me know. Bill Hayhurst

An axe is a legacy tool from the Stone Age. There are as many applications for Voice Mail as there are stars in the sky. Sometimes you just need an axe. Bill H

I have searched the manual and Wiki and I can't find a Voice Mail Broadcast Message feature. Is there one? Its a common message that is usually left by a manager and is sent to a list of mailboxes. Also, how about a message forwarding feature that will automatically forward a message from one mailbox to another after a period of time? Maybe even delivery of a message to a cellphone just after the caller hangs up? Bill H

I only have one Domain. (localhost) The Trunk rings in to Ext 105 I am using the most recent Version of pbxnsip Has anyone else tried this? Let me know if you have any success. Bill H

It still does not work. I entered 16032838080 in the Domain Address Book and Named it Test 1 for First Name and Test 2 for Last Name. I called in and the display reads: 16032838080 Unknown Name This line is from the pbxnsip Call History related to that call: 2007/07/04 09:33:25 16032838080@66.193.176.35 Bill H (105@localhost) Am I doing something wrong here? What is next? Bill H

I added my cellphone number and a name to the Domain Address Book. I then dialed into the pbxnsip and all I see is the number of my cellphone. What should I see? I was a bit confused by your last post. Bill H

In the PBXnSIP file there is a folder for the Recordings. Does anyone know of any external search software that will search the the Recordings folder looking for files by different criteria. Example: Search by Date Search by In or Out call Search by Number Dialled or Caller ID Thanks, Bill H

Unless I am am mistaken, the Address Book works off the Caller ID Number. I am not concerned with Caller ID but rather the actual Trunk that the call came in on. 15 different registrations with 15 different trunks. Is it possible to identify which trunk the call came in on? Bill H

I have 15 different incoming 800 telephone numbers and want to be able to identify which one is ringing in at any given time. This will allow the receptionist to answer the call with the correct greeting. I don't want to use 15 seperate buttons, I do want to use the LCD Display on the phone. I see in the Trunk setup where it speaks of a "Display Name". This name does not apply to the SIP Registration. I put in a name like "PBX 1234". I did make some test calls in on the 800 number assigned to this trunk. I looked in the PBXnSIP Log File (set to show all) and did not see my test name anywhere. I also looked in the CDR and did not see it there either. So my questions are: Is it possible to identify which 800 number is ringing in? Is the Trunk Name (Display Name) available anywhere at any time? Thanks, Bill H

How long, in seconds, is the voice announcement? How important is it to start from the beginning? Bill H