adampcpi

What costs are associated with moving from PBXnSip v4 50 Extensions to the new v5?

Is there an upgrade path from PBXnSip? We are running a 50 license of PBXnSip and looking to upgrade to take advantage of some of the newer features.

Is there a way to send an email upon someone entering a agent group? What I am looking to do is create an alert when a new caller enters the queue via email or some other system. They may not always be at their phones to hear it ring but a email to sms message would work for this particular need that alerts them that a call is waiting for them. I was going to build a program to monitor the active call list but using the url method doesn't seem to be stable to constantly poll it. -Adam

We keep having this issue where the first DTMF tone that is sent is not being picked up. We are running the latest version of PBXnSIP and using 4 Patton SN4114/JO/EUI devices to convert the PSTN lines for our phone system... Not sure if this is a Patton issue or a PBXnSIP issue but one thing to note is I have noticed that when placing an outbound call the outbound call is definatly quieter than placing a extension call... It just seems to be missing the first digit of the extension people are trying to reach and they get the "This extension number does not exists" and they have to dial it again.. Example here is a caller dialing into our system and trying to dial 3018 [8] 2011/07/15 12:20:55: Packet authenticated by transport layer [7] 2011/07/15 12:20:56: Received RFC4733 DTMF on codec 101 [6] 2011/07/15 12:20:56: Received DTMF 0 [8] 2011/07/15 12:20:56: Packet authenticated by transport layer [6] 2011/07/15 12:20:56: Received DTMF 1 [8] 2011/07/15 12:20:56: Packet authenticated by transport layer [6] 2011/07/15 12:20:56: Received DTMF 8 [8] 2011/07/15 12:20:57: Packet authenticated by transport layer [8] 2011/07/15 12:20:59: Last message repeated 11 times [8] 2011/07/15 12:20:59: Attendant: Timeout (press) [8] 2011/07/15 12:20:59: Play audio_en/aa_not_existing.wav space20 [8] 2011/07/15 12:21:00: Packet authenticated by transport layer [8] 2011/07/15 12:21:03: Last message repeated 7 times [8] 2011/07/15 12:21:03: Play space20 [8] 2011/07/15 12:21:04: Packet authenticated by transport layer [8] 2011/07/15 12:21:05: Last message repeated 2 times [6] 2011/07/15 12:21:05: Received DTMF 3 [8] 2011/07/15 12:21:05: Packet authenticated by transport layer [6] 2011/07/15 12:21:05: Received DTMF 0 [8] 2011/07/15 12:21:05: Packet authenticated by transport layer [6] 2011/07/15 12:21:06: Received DTMF 1 [8] 2011/07/15 12:21:06: Packet authenticated by transport layer [8] 2011/07/15 12:21:06: Last message repeated 2 times [6] 2011/07/15 12:21:06: Received DTMF 8 [8] 2011/07/15 12:21:06: Call state for call object 1173: alerting Any help with this would be appreciated...

We are having issues with the multicast paging being stuck on... According to all my users they are only making 1 page... They press the page button wait for the beep and talk then hang up their phone... What then happens is a empty page gets sent out that puts all my snom 360 phones in PA mode and it is completely silent... When this is reported to me I go into the Current Active Calls list and I see the page... I hit the X button to end the call and it immediately starts playing all the pages that have happened since it was playing that blank page... I was hoping to script a utility to see how long the page has been playing and then automate ending it but I have noticed that the date/time of the page is displaying bogus information (see attached pic)... In our environment we have to use multicast and we switched to the delay to avoid feedback into the paging system from other phones... Version: 4.0.1.3499 (Win32)

I don't think there is a way to do it through the interface but yes you can script it.... It takes a little bit to setup but once you get it setup you just have to have something hit a specific URL at the times you want it to change.... You can get all the required information from by viewing the source of your current dial plan... Example with my setup here is an example URL that would configure all my trunk lines: http://admin:password@pbx.com/dom_dialplan_edit.htm?dialplan=1&name=Standard+Dialplan&global=true&dpz0=0&dpa0=100&dpb0=&dpc0=&dpd0=&dpz6=13&dpa6=24&dpb6=%2B&dpc6=%3F%3F%3F%3F&dpd6=&dpz2=9&dpa2=25&dpb2=%2B&dpc2=8%3F%3F%3F%3F&dpd2=&dpz1=8&dpa1=50&dpb1=7&dpc1=*&dpd1=&dpz3=10&dpa3=51&dpb3=10&dpc3=*&dpd3=&dpz4=11&dpa4=52&dpb4=9&dpc4=*&dpd4=&dpz5=12&dpa5=54&dpb5=11&dpc5=*&dpd5=&savedial=Save&dialed=&dialresult= So to break it down for one of my trunks: &dpz5=12&dpa5=54&dpb5=11&dpc5=*&dpd5= That currently has a Pref of 54 and is Trunk ID 11 and the Pattern is * Lets say I wanted to make it Pref 49 to I would just have to change the 54 to 49: &dpz5=12&dpa5=49&dpb5=11&dpc5=*&dpd5= Hope this helps... This is probably not by any means an official way to do it but it works -Adam

Right now I can see the calls in the active call list to the paging extension so why can't this be included in the call logs or any other internal call?

I am using VB so what I did was just pass the string twice but putting the admin username/password in the URL: http://admin:pass123@mydomain.com/ajax.htm...;token=whatever $call_index = $xml->Calls->Call->Index; // Not sure what this is This is the call identifier which now that I know I can pull this I am planning on building a front end for our receptions to force pages to stop paging... (ie you can pass that ID to http://admin:pass123@mydomani.com/dom_call...lete_call={ID}) I was looking for a way to do this without giving the receptionist full access to the PBX system. $call_gain = $xml->Calls->Call->Gain; // Not sure what this is either Mine never showed anything in Gain put typically means something do to with volume amplification... -Adam

I tested it here and that information may be handy... the authentication may be a problem though... in my brief testing you have to run it twice for it to work... once to login and then a 2nd time to get the information...

It still has to go through the PBX somehow as the users are dialing the paging extension then it records the message and the plays it back to the RTP stream... in my current setup the call logs don't even display internal extensions so maybe by enabling that I can see dialed to the paging extension... Is there a way to include internal calls in the call log and not just external calls?

We are using the record option but the issue is trying to kill the long page... haven't found a way to do it yet... the call does not show up in the Current Active Calls and I don't see it in any log file... How can I log all calls to the paging extension?

We just went live with our pbxnsip setup and we are using paging using multicast and on a delay to avoid feedback from nearby phones... we have had cases were people accidently hit the page button and then forget about it and then they hang up... then it sends the page... the only way I have been able to get rid of this is go in the console and end the call... Question 1: is there possibly a way to end the current page using a feature code? Question 2: is there anyway to see the extension paging on the phones? Question 3: is there anyway we can limit the length of pages? TIA -Adam

Scenario: I was on my Snom 360 phone and someone tried to call my extension and it was busy so they got the prompt "press 1 to receive a call back when the extension becomes available".... when I hang up my phone I assume that person got a call and then my phone started ringing but it said I was calling myself... shouldn't the caller id information be the person that called in the first place? Just seemed really odd to not know what person is calling you when using this feature...

Is there a way to set the default settings for snom phones in pbxnsip so when we do a factory reset that the settings are pushed to it (ie. default ringer, custom volume levels, dialtone on hold = false)....

Not sure if this is a bug or not but the icon on the WAC screen does not change for me when you login/logout of the Agent.... it works for DND though... In my testing I had one computer pull up the WAC screen while the agent was signed in and it had the checkmark icon... I then signed out the Agent and the checkmark icon still displayed... I then used another computer and logged into WAC again and the checkmark icon is not present... If I do a manual web page refresh it corrects itself but would prefer it would operate just like DND does... Is this a known issue? -Adam

Thanks... found the setting and adjusted it...

Trying to phase out our old phone system and our old system has a console that shows the status of each line so that the operator knows if it is busy or not. Using the web based console (WAC?) will work but it seems to timeout after a while and the user has to log back in.... Is there a way to adjust the timeout on this so that it can at least work for a 10 hour shift? -Adam

Got a puzzling issue.... I have setup a bunch of Agent groups and my Extension is 3014 and for some reason it keeps adding me to all the Agent Groups after I have removed myself from the groups... I have confirmed I did remove 3014 from the agent list but it seems when I sign off and back on using code *64 it adds me back to ALL groups.... pbxnsip v4 and Snom 360 phones... -Adam

I have the agent queues working properly and all seemed to be configured how I want them but can't seem to get the button to login/logout of the queue... When I use the feature codes *64/*65 it works and turns off the light on the button that I have configured for "Agent login/logout" with the parameter of the agent queue number in this case "4002"... Is there something else I have to do to make the button simulate the *64/*65? Using pbxnsip v4 and snom 360 phones... TIA -Adam

I added the supplied Action URL's to the DND Action URL's and it works with the DND button.... Thanks! Now is there anyway to change make this setting the default for all phones and push it down to the phones via pbxnsip or do we have to log into each phone and make this change? TIA -Adam

We are running pbxnsip v4 and using Snom 360 phones. Paging is working in multicast mode just fine but some users want to block pages due to being on another phone or don't want it to be on their phone just temporarily (so I don't want to configure their phones to never page). I have tried turning DND on but the paging still goes through. Also on a side note the only way we have been able to turn the volume down for the paging is to turn the speakerphone volume all the way down but when you want to use the speakerphone you have to turn the volume back up and then back down to turn the paging down again. Any work around on this one? TIA -Adam