Bill H

We have entered our cellphone numbers into each extension account. Normal calls coming into PBXNSIP are answered by our Auto Attendant. When we call into PBXNSIP from our cellphones the Auto Attendant gives the choices to make a call, go to our mailbox or connect to the Auto Attendant. We would like to use the "Call Monitor" feature from our cellphones. Does anyone know the sequence to do this? Thanks, Bill H

As we already know, Metropolis can't show any ACD info until it receives it from PBXNSIP. I can not find a way to have PBXNSIP send any info about an incomming call answered by an ACD. The $e string inserts the Extension number but not the ACD number in the CDR. Of course the ACD is not an extension but rather an Account. If I remember correctly, the older PBXNSIP CDR output would send (by UDP) the Account Number instead of the Extension Number. I was able to work with that. But now the Account Number seems to been gone. So the question is --- What String Character is used in the XML CDR String to get the ACD (Account Number?) that answered an incomming call??? Thanks, Bill H

We are using the Metropolis.com Office Watch CDR Call Accounting software. It runs on the same server as PBXNSIP and receives its data from PBXNSIP via an IP Port It can also display the Raw Data it recieves as a troubleshotting tool. We currently use the string $w$5e$10c$5d$o$x$y which sends the following: Date Extension Caller ID of Remote Party Duration of Call IN or Outbound Direction of Call Originating Trunk Destination Trunk This works well for calls to Extensions. We also have ACD Groups that answer calls when they come in on our T1 Gateway. I have tried all the possible $(String) options in the WIKI and using the Raw Data Tool, can not seem to see any ACD related data out of PBXNSIP. The ACD's are set up as Accounts in PBXNSIP as are the Extensions. Does anyone have an idea as how to get the ACD info???? Thanks, Bill H

I would suggest that you use the RESTORE FACTORY DEFAULT SETTING feature. Then upgrade to the latest firmware from Grandstream. It sounds like you either have a defective phone or there is a setting (CODEC, SRTP etc) that is causing the trouble. Default the phone then add only the basic information to make it operational. If it still does not work, try and register it to another service. If it doesn't work there then it looks like it might be defective. Your trouble sounds like it is either Hardware, Firmware or Settings. Bill H

Well Joe it looks like you have subscribed to a SIP Service Provider already. (ITSP) Go to the trunks settings of the trunk and set the last entry "send call to extension" to a destination where you want the incoming calls to go to. Use an extension first, it will be easier to see if it works and troublshoot if it doesn't. Then go to the "Dial Plans" category and create a new dial plan and give it a name like SIPTrunk. For a beginner I would suggest a simple dial plan like this: Pref Trunk Pattern Replacement 100 SIPTrunk 9* * Now you will dial 9 first followed by the telephone number you want to connect to. Don't forget that you need to dial the entire number as you would from a regular telephone. Example 1 800 234-5678 Or something different if you are outside the USA. The number of channels you receive from your SIP Provider will determin how many calls can be placed at one time. Once you get it working, you can move on to Auto Attendants, ACD's and IVR's for call distribution. Bill H

OK I see. You didn't mention a second AA in your previous post. I believe that all of the AA's use the same (common) prompts for the "hard coded" responses. I guess the only way to make it work would to be crafty and arrange your AA's in such a way that prompts make sense no matter which AA is delivering the prompt. Bill H

I can't think of anyway to send a different Ring Tone to a phone based on the AA selection (1 or 2) where they both go to the same destination. (Ext 20) You could put a second extension registration on the same phone like Extension 21 and possibly have it ring different. Then have the AA direct Networking calls to Extension 20 and Sales calls to Extension 21. You can record prompts and save them in the audio_en (or audio_?? for Dutch) folder. Just go through the prompts and look and listen for the one you want to change. Then change the existing prompts' Name to something a little different so it isn't rcorded over. Then name your new prompt with the name of the original prompt. The recording must be in 8 Khz Mono I believe. Check to Wiki to be sure. Hope this helps. Bill H

Sorry, I meant to say my previous posts are in the SOAP section of this forum. Bill H

I worked with the SOAP and IVR several months ago and was unsuccessful. Please if you get PBXNSIP to respond to the SOAP request let me know how. See my posts in the IVR section of the forum. Maybe the newer versions have changed something. My external server echoes back the Call ID with the Extension number but PBXNSIP says the call does not exist ????? Bill H

No I did not have that turned on. I have turned it on and it now sends the notification OK. Thanks, Bill H

In the Extension Settings I have: Log Register Expiry set to: Email to Adm. and User but no notification is sent when the selected extensions' registration expires. It did work a month or two ago and stopped. Is there any other setting I may have changed that affects its operation???? Note: The extensions Email address is correct. I do rcv Voice Mail notifications and attachments OK. Bill H

Anyone have a clue on this one??? Bill H

At this point it looks like we need some info from the phone and/or the PBXNSIP PBX. In the SDP (Session Description Protocol) portion of the SIP message the phone tells the PBX where to send (IP address) RTP (Voice Packets) on which port. This is also true for the reverse, with the PBX telling the phone where it is and what port to use for voice. In your phone see if there is Syslog feature. Download KIWI Syslog, get your phone to send some logs to it and that will get you in the ballpark You may not have a port problem but rather an IP Address problem. The PBX and the phones need to send their Public IP addresses ex. (34.22.123.201) in the SDP portion of the message, not their Private IP Addres. (ex 192.168.1.100). The connection and ringing portion is easy since your phones are registered to the PBX. The next step is for the PBX and phone to agree on a Codec (Use a-law or u-law for test) You can also use the Logging feature in PBXNSIP. Set it up for level 7 and you will see all the messages sent. You will learn more about SIP when you are done. Bill H

I can think of 2 ways to do this. One works well and the other has a little side effect to it. Method 1. Side Effect If the caller directs a call to an extension by dialling the extension number from an Auto Attendant, then the call will go directly to the extension. The only way I can see to place a "Please Hold while we connect your call." then give the person a 5 second advertising is to record it in the Ringback.wav file along with the ringing sound. The extension will ring while this message is being played. This will be a common message played everytime an extension rings. The side effect is that when you make an extension to extension call, you will also hear the message too. You may also hear this message when you make outside calls. This depends on your gateway or SIP Trunk use. Method 2. Works Well Don't use extension numbers in the Auto Attendant, use single digits. Using a single digit will allow you to control and send the caller almost anywhere in PBXNSIP. You will need to create an ACD Group for each extension that is involved in the advertising response. You will have one extension in each group. A little high on license usage but it works. I have tried to use IVR Groups with a SOAP program but PBXNSIP says the call does not exist. Thats another story. So send the caller to an ACD Group as listed above. Place your message and advertising in the ACD Group recording. The message will play and then the caller will be transfered to the extension. Since the caller is sent to an ACD Group you can have a different message played for each extension. Hope this helps. I tried to attach a wav fille so you hear what I mean. Bill H

Well first I am going to re-wite your request a little differently. 1- Have the office secretary extension be the first to answer. Easy. Put her extension number in the Trunk where it says "Send call to extension:" This will send incoming calls to her extension. In "Redirection" set Call Forward No Answer on her extension to a newly created ACD Group. In "Redirection" set the Call forward no answer timeout: to 10 seconds or so (try to get 2 rings). ---------------------------------------- 2- After 2 rings (10 seconds or so), play a message like, “Thanks for calling….” During that message have the ability for other extensions to pick up the call OR have a couple other extensions ring for pickup. The "Thank you for calling..." message would be recorded in the above new ACD Group. In the ACD Group setting "After hearing ringback for (s) ..." enter a 1 In the ACD Group setting "... redirect the call to the destination (e.g. "73"):" enter a newly created Hunt Group number. The above items will answer the call, play the message, wait 1 second and then transfer the call to a hunt group You asked "During that message have the ability for other extensions to pick up the call OR have a couple other extensions ring for pickup." I don't think the first half of your request (2) is possible. Anyway, the caller will only be in the ACD Group for the lenghth of the greeting (plus 1 second) and no one will even know that there is a call in there. The second half of your request is done with the Hunt Group. Add the extensions that you want to ring in the "Stage 1 Extensions:" area and they will ring as soon as the message (plus 1 second) is finished. Set the Duration time for how long you want them to ring. ------------------------------------------- 3- If no other extension picks up the call, have the call sent back to the secretary extension where the caller can leave a message. Add 8+operator extension number as the "Final Stage: in the Hunt Group ------------------------------------------- Get fancy and "Name" the Hunt Group "No Op Avail" In the Hunt Group "From-Header:" select "Group Name" That should put the name "No Op Avail" in the LCD Display of all phones that ring in the Hunt Group. Hope this helps. Bill H

Like everything else, there is a setting you need to look at here Under Admin / Ports RTP:* Port Range Start: 49152 Port Range End: 64512 This establishes the RTP (Voice) port range to be used by PBXNSIP. Also set the Preferred CODEC to at least 0 8 18. Make sure that PBXNSIP is assigned as a Private Static IP Address to your router. Set port forwarding to the Private PBXNSIP IP Address to the range of RTP and also 5060 to say 5100 for SIP signaling. Also, check to see what Codecs the phones are using. What type of phones are you using??? Hope this helps. Bill H

I am trying to get an IVR Node to dial a telephone number (12122351234). My pattern is !1!12122351234! The prompt comes back and says I am not allowed to make that call. The Wiki says "The destination may be any dialable number. If the number requires a dial plan, the default dial plan of the domain will be used." When I look at the Call Log it says there is no dial plan assigned. What am I missing here? Also, there is a Dial Plan in the Auto Attendant, but it doesn't work either. When I try using say 1 to dial a telephone number (12122351234). It returns the same prompt. I am not allowed. Bill H

I have worked with the MVP-210 unit. It takes a little while to understand how to configure it. It is however very straight forward. It was designed from the ground up as an "Off Premisis Extension" adapter for traditional TDM systems that have an available analog station port. You install one unit at each location and they communicate peer to peer without registering to each other. Works well if you have a Static Public IP Addresses at each end. Going through routers is a little more difficult. It has 2 ports that can be individualy configured as FXO or FXS. There are more flexible gateways out there like AudioCodes and Grandstream. It has one distinct feature using what they call a "Telephone Book" (I think thats the name). You could have many of the MVP-210's (or SIP phones) at remote locations and be able to call any of them from the one at the PBX while only using one trunk account. Its kind of like and intercom system within a PBX of its own. Considering the price and the learning curve, I would stick with the two others I mentioned earlier unless you have a special need for the MVP-210. Bill H

----------- We have 4 gigabytes in the server and are not running any other programs on a regular basis. We are running the needed processes for Windows to operate. Could it be a memory Leak? Accumulating memory over a week and then goes bad? The Task Manager says it is usin 84.016K of memory for PBXNSIP. Is this the method you suggest or is there something better? Task Manager also reports over 16 million handles, that seems high to me. Bill H

We have been using PBXNSIP for about 9 months now and have never experienced a system crash. Version: 2.1.6.2433 (Win32) and also when we used the previous version. 2.1.5.???? Everything else in the server works OK during the crash and the PBXNSIP service re-starts again OK. Since 1/3/08 we have had several crashes as indicated in the list below. Type Date Time Source Category Event User Computer Error 1/3/2008 09:34:17 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/31/2008 09:48:51 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/31/2008 10:24:44 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/11/2008 10:32:54 AM Service Control Manager None 7031 N/A PBXNSIP Error 2/4/2008 10:34:08 AM Service Control Manager None 7031 N/A PBXNSIP Error 2/4/2008 10:43:59 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/5/2008 11:00:10 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/30/2008 11:06:43 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/11/2008 11:24:39 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/5/2008 11:29:58 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/18/2008 02:36:33 PM Service Control Manager None 7031 N/A PBXNSIP Error 1/4/2008 04:15:35 PM Service Control Manager None 7031 N/A PBXNSIP Error 1/31/2008 04:17:22 PM Service Control Manager None 7031 N/A PBXNSIP Error 1/18/2008 04:32:31 PM Service Control Manager None 7031 N/A PBXNSIP There seems to be a pattern based on the cluster of failures at different times, but I don't know if this means anything. I looked at the pbxlog, but it isn't time stamped so I could not get any meaninful info from it. Is there a way to get the last message sent before it crashed? Any ideas as to why it fails? Bill H

Well there is good news an the horizon. I have been working closely with PBXNSIP concerning the long sought after console. I strongly beleive there will be a Beta Test Release available around the end of January 2008. Bill H

Is it possible to have a caller that is in a Mailbox enter a digit that will transfer them to the Mailbox owners cellphone? I tried the cellphone number in the Mailbox Escape Account and it did not work. I tried *77 + cellphone number and it did not work either. I tried using the Mailbox Escape Account and sent the caller to a dummy account where is was set to Call forward all calls to: the cellphone and that worked OK. It just uses up an additional account. Any ideas or is it just not possible. Thanks, Bill H

Does anyone have a list of Windows Services that can be shutdown (Disabled) without affecting PBXNSIP's operation. Thanks, Bill H

Is there a URL to download the newer 2.1.5 version? Bill H

We normally receive a New Message Indication on our Aastra phones along with the total of new messages in the Mail Box. Our office was closed for a week and all calls went to their respective extensions OK. Some of the more active extensions were up to around 40 or 50 messages. All messages were checked and deleted. Now, all of a sudden, around the same time every day (2PM EST) almost everyone gets a New Message Indication with around 80 to 130 new messages. These are all Blank/No Message. Anyone have an idea as to what this may be? Thanks. Bill H