Pradeep
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Posts posted by Pradeep
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Aha, that works - do the "Agents" & "Calls" stats refresh automatically or "push" AJAX style or is it a case of pressing F5?
Currently it refreshes automatically, every 5 or 10 seconds
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I have a demo today and I need to know how to have a group of phones be able to monitor an incomming trunk so that everyone can see when the call comes in. I have searched the Wiki and found nothing that relates to this topic in particular. I greatly appreciate any help.
Tom
Does this wiki link help http://wiki.pbxnsip.com/index.php/CO_Lines ?
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One of our customers is having an issue, since upgrading to 3.x. This is on a server hosting multiple customers.
Here’s the exact scenario:
1. A call comes into the office and it needs to be transferred to an agent group that handles that type of call. For instance, we tried this earlier from 123-456-7890 (cell phone) to our main number (234-567-1234) auto attendant and dialed an extension. 119.
2. We transferred the call to x176 (our certificate agent group).
3. There are two agents that are on the agent group, extension 105 and 146. Both phones ring when the call is transferred, which should not happen. If extension 146 tries to answer the call, there is no body there and x105 continues to ring. When x105 answers the call, then the person is connected successfully.
Any ideas?
Thank you.
Looks like this is similar to the post http://forum.pbxnsip.com/index.php?showtopic=1203&hl=
Could you please download the version specified in this post and verify?
thanks
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I have a client with this same issue! he is setup with pnp with a snom 360...when he hits the DND button on his phone it changes the status on the pbx to do not disturb but when he turns it off it does not turn it of on the cs410.
what can i do to rectify this?
pls advise asap
Can you please verify the PBX log file to see whether PBX is receiving *78 and *79 codes during the DND ON and OFF operations? Alternatively, you can also verify it in the SIP trace of the snom 360
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The call from sip:1732961****@sip.jivetel.com;user=phone to sip:1800624****@sip.jivetel.com;user=phone has been disconnected because of media timeout (120 seconds), 13071/19057 packets have been received/sent
What does this mean????
This generally means that the PBX disconnected the call because there was no media for 120 seconds
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Thank you for your speedy response.
where do i put this string though? in the user account setting on the phone i am registering or somewhere in the settings of the extension? pls advise.
You can add the static registration under "Registration" tab of the extension. At the bottom of the page you will see "Add Contact", type in the contact as described and hit "save"
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how can i setup a static registration? i have the same issue where the person on call sometimes has to step out of the office for a few minutes and needs the calls to simultaneously ring on her cell...pls advise the workaround asap.
Checkout this wiki link http://wiki.pbxnsip.com/index.php/Manual_Registration
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Hey there. I've been setting up an OCS2007 with Exchange UM environment with success thus far with routing calls between SNOM320 sets and OC clients. I'm now looking into the UM portion, specifically the Auto Attendant. I can call in, receive the greeting and ask to be routed to an extension. Once it dials the extension the phone & communicator ring, which is all fine. My issue is that if I hang up the phone while the AA is calling the extension, the extension keeps ringing and then forwards to voice mail. How can I get the auto attendant to terminate the call when the outside caller has ended the call?
Can you please get us the PBX log (with the SIP logging enabled)? or the wireshark network trace will be okay too.
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2008/09/19 09:58:11: Starting up version 3.0.0.2998
[[5] 2008/09/19 09:58:17: Trunk 1 (PSTN) has outbound proxy udp:127.0.0.1:5062
This is not a problem. There are some translations missing for non-english languages. Just some debug info to see what is missing. We will remove them from the default output.
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I just purchased the CS410, and while verything was a breeze to setup and get working, I am having problems with it properly detecting PSTN calls being terminated.
Maybe because of the different tones our country uses, so I would like to request help from anybody setting up the proper values.
Country Tone Freq/Hz Cadence/s Costa Rica Busy tone 450 0.33 on 0.33 off Costa Rica Call waiting tone 450 0.15 on 0.15 off 0.15 on 8.0 off Costa Rica Congestion tone 450 0.33 on 0.33 off Costa Rica Dial tone 450 continuous Costa Rica Ringing tone 450 1.2 on 4.9 off
Using 3.0.0.2998 (Linux)
Thanks in advance.
Have you already checked this page http://wiki.pbxnsip.com/index.php/Installi...N_gateway_setup ?
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Hello every one.
Here is another PBXnSIP problem.
PBXnSIP support has never actually helped fix any problem so far, so im hoping that other unfortunate system users / builders can help.
System Config Details
Operating System. Redhat Enterprise Server REL 5.0
PBXnSIP version. 3.0
Scenario.
Incoming calls go to an Agent Group and into the queue.
Announcements and music on hold work OK
The queue passes callers through to the agent in the correct order OK.
Note: there is only (1) agent
Problem Description
If the "agent" is not attending the phone and the call is past to that extension the phone rings continiousley IE: non stop and
WILL NOT GO TO VOICEMAIL.
Things I have tried including:
1)
From the "Agent Group" Level.
A) If the caller already waited longer than (s) >>> set to 5 seconds
Redirect to the destination >>> EG: 73
Result: agents phone keeps ringing. the call does not redirect.
Comment: Would not be a good fix any way as all calls in the queue would be redirected/
2)
From the Extension level in "Redirection"
A) Call foward on no answer to : EG 73
Call foward no ansertime out: EG 5 seconds.
Result: agents phone keeps ringing. the call does not redirect.
NOTES:
If a system extension calls that agent direct IE: not VIA the QUEUE, calls go to voicemail as normal.
After hours: All calls go directly to that same agents extension IE: not VIA the queue. Calls go to voicemail as normal.
Commment:
I have been in IT for 20 yaers and have never seen a product (Sold by a reputable wholesaler) with as may bugs and as little support for resellers as PBXnSIP.
It seems that this application has some powerfull features but it is extreamly weak when it comes to basic features, stability and functionality.
The agent group redirect(Field: If the caller already waited longer than (s)...) is applicable if all the agents in the queue are really busy. Ex: If the group
has 2 agents and both are on the call then, the for the 3rd call (new incoming call) this settings apply. That means, after the waiting interval, say 20 econds, the call will be redirected to the redirect destination.
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If I add '!^E$!156!' would that match and loop back to extension 156 or do I need to have the DTMF input in there to make it trigger?
I am not sure I followed the issue completely. But if you are looking for how to set the DTMF match list on the IVR node, there is example under http://wiki.pbxnsip.com/index.php/UHLL_Interface#Maid_Codes.
With match list shown in the example, pbx collects digits until you press the pound (#) key.
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yea thats what i did, sorry.
except i was using a \ not a /
so that got rid of the error, however it still is not working.
I can view the file manually.. ie ... http://localhost/img/logo2.gif
however every time it loads http://localhost/img/i11.gif
Using the help of http://wiki.pbxnsip.com/index.php/Changing_the_Appearance, I just tested this. It seem to work fine.
(if you keep the logo2.gif under html/img folder, please use html/img/logo2.gif on the "Appearance" page). Make sure that you clear cache on the browser. Otherwise the changes wont show up.
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BTW incoming calls through the same trunk kept working all the time
After some days of testing i gave up, back to 2.1.12.2489.....
I am not sure whether you have already tried setting "Accept Redirect" flag to "yes" in the trunk settings page.
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I'm assuming this is Intel only but:
The installer does run in ppc, maybe it is possible to check if the platform is ok before installing?
Or make it a UB ;-)
Currently, this has been tested only on Intel. BTW, what is UB?
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Currently, it is done for a reason. But in the future releases we will make it settable.
We have posted the version 1.9.2.22 that has a configuration setting for the popup on the 'Settings' tab. With this you can turn it on or off.
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I am trying to set a codec preference on an extension and when the submit is hit the preference disappears? BUG?
IE: 0 8 9
On the latest software I tried setting the codec preference on an extension and it seem to work fine. What version of the PBX are you running?
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Hi Pats,
thanks for answer...how can I check it from the SIP traffic?
Thank you.
He might have suggested to check it under the trunk configuration.
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hi pradeep,
just installed pbxnsip on mac. works good so far.
any idea when you will be using my installer scheme with launchd and install path /Library/Pbsnsip?
thanx
christof
We have plans do that on the next pbxnsip software update.
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I notice in one of the latest releases they added creating accounts from CSV files. I cannot find any documentation on how this works, where do we find it.
We have added a new section in the wiki for this purpose. Here is the link to the section
http://wiki.pbxnsip.com/index.php/Creating...sing_CSV_import
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Maybe it would be worth mentioning that it's usable from 9.0 and up?
Mac X is almost 7 years old by now.
Well, the target was built using the latest MAC version, v10.5 Leopard. But you should be able to run it on v10.4, Tiger.
More on the MAC versions http://www.opensource.apple.com/darwinsource/
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Yes...yes. We have released the MAC version of the PBX. It is available on our software download page under "Mac OS (Darwin 9.0)" name.
http://www.pbxnsip.com/software
On the software page I found the installer for Mac 9.The news page states that an installer for Mac X has been released, but it is yet not on the downloadpage.
Has the software been made available yet?
Note: We aren't using Mac X ourselfs, but I noticed it while getting the windows version
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Latest 3.x version(3.0.0.2298) should has fixed this problem
Javascript validation did not consider the :port option. We will fix it ASAP.
3.1 Beta Image available
in New Features/Versions
Posted
You can grab the latest and greatest here - http://pbxnsip.com/download/pbxctrl-3.0.1.3023.exe