Tim
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Posts posted by Tim
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Hi,
We have a client that wants the custom Auto Attendant message to replay after the caller enters an invalid option. Currently we are stuck with the only options being dead air after the "you have entered an invalid extension" or the system "Please enter an extension number" prompt.
Neither of these options are good options, since if the person entered the wrong option initially, they probably don't know which option was the correct one to enter. Why can't we replay the custom greeting after an invalid option message?
My only thought is to create an IVR account with an "You have entered an invalid option" message and send the call back to the AA. Will this cause problems with the loop avoidance technology in the system?
Thank you,
Tim
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http://pbxnsipsupport.com/index.php?_m=kno...kbarticleid=392 talks about the change in the logo size for v4. Also, in the very recent versions we have made both the operator console & the web interface to use the same logo.
Thank you for the link with the new logo size. FYI, the Version 4 Administrator Guide has the old logo size in the white labeling page.
Tim
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Hi all,
We seem to be having 2 problems with the white labeling of our pbxnsip 4.0 server. The first problem we are having is on the new web operator console. The footer values are being white labeled correctly (Copyright, etc) but the logo is still the default pbxnsip logo. Are there additional options in the branding that need to be enabled to make the logo show on this page?
Also, in the main User and Administration pages, the logo is being tiled across the top of the page. The logo we are using is 120x30 pixels as specified in the Administrators Manual. Is this the correct size logo?
Tim Donahue
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We have a client with a PBX (3.4.0) that is using the hot desking feature for their spare phones. When they use the hot desking, the MWI does not come on when they have new voicemails in their mailbox. I thought we had tested this in a previous release and the MWI had worked when we tested it.
Has anyone else noticed this behavior?
Tim
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We are looking for an ATA with support for TLS and SRTP. Can anyone make a recommendation?
Thanks,
Tim
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Looks like we do not have added that to the mac address range for the cisco's mac address range in the pnp.xml. Temporarily, you can keep the custom pnp.xml with your mac address added to the range and restart the PBX service. Let's know if you do know have a copy of the pnp.xml
I have a "pnp_parms" directory with a bunch of .xml files in it, but I do not see a pnp.xml file. Is there a way to make the system spit this file out or where can I download it from?
Tim
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Hi all,
I have a Cisco 7960 phone that I am attempting to provision through the Plug and Play functionality of pbxnsip. I am getting the following error, is there anything I can do to resolve this or is this information hard coded into the pbxnsip software? We are currently using version 'pbxctrl-centos5-3.4.0.3201'.
[8] 2009/08/31 22:02:54: Could not identify phone vendor for MAC 00146A4D331A
Thanks,
Tim
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From my experience, it depend of the phone itself. Linksys/Cisco yes you type http://xxx.xxx.xxx.xxx/advence/?reste
other phones... I have no clue.
The problem is not resetting the phones. I know how to reset the phones to factory default, the problem I am trying to solve is finding an easy way to change the programming of the phones. We are currently trying to do this with a Snom phone. Here is an example of what I am trying to accomplish, I want to take the phone provisioned for extension 120 and changing it so it is provisioned for extension 125.
As previously stated, this was easy when the programming was based on the MAC address, you removed the MAC address from extension 120, added it to 125, and rebooted the phone.
Is there a procedure that we can use to change the extension the phone is provisioned to, without actually interacting with the phone once its been setup to be provisioned from the pbxnsip server?
Tm
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Couldn't you use the username based provisioning?
and delete the old account?
(of course email them first ;-)
matt
We are not removing the old account, just changing the extension that the phone is associated with.
We already use the username based provisioning, which is turning out to be the problem. When the provisioning was based on the MAC address, changing the MAC address to a new extension would reprogram the phone with the new extension. Now somehow we have to change the username that has been programmed into the phone.
Tim
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We are currently running 3.3.1.3177 on CentOS 5 for a hosted PBX for one of our clients. We are using the PNP configuration for the phones and have a phone that we need to reprogram for a new extension. Short of having the client send us the phone for reprogramming, is there a way for us to change the account the phone is associated with?
Tim
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Hi all,
I have a system that I was looking at upgrading to 3.4.0.3201 later this week, this system is currently on 3.3.1. Does anyone know if there are any changes that I will need to go back in and make to the accounts after the upgrade, or will this be a seamless upgrade where everything "Just Works" after I am done?
Thanks,
Tim
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We are in the middle of a staged rollout for a client, and we had to update to 3.3.1 to resolve some voicemail problems the client was experiencing. Now when we try to auto provision the phones, the PBX is not providing the provisioning information. I found the thread explaining that we need to setup the HTTP Client User and Password in the phone, but when it attempts to pull the config, the setting-files parameter is blank.
<?xml version="1.0" encoding="utf-8" ?> <setting-files />
Can anyone explain what the problem is that is causing this, and how we can resolve it so we can ship the next batch of phones that are due to go out today?
Tim
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which is the current version of software you are running? We are running 3.3.1.3177 (http://pbxnsip.com/download/pbxctrl-3.3.1.3177.exe) and do not see this issue.
We are running pbxctrl-centos5-3.3.1.3177
Tim
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Maybe that trick can help: In the ANI field, you can tell the PBX to use a different ANI when using a specific trunk.
For example, "9787462777 Trunk1:123" would mean: On trunk "Trunk1" use "123", otherwise use "9787462777" as ANI.
Is it possible that in the future an option in either the 'Remote Party/Privacy Indication' or the 'Rewrite global numbers' field, I am not sure which this would be applicable to, that sets the trunk to 'Send extension', instead of sending the ANI?
Tim
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Alex,
We have used the Grandstream GXW-4104 Gateways with good success, and I believe they make an 8 port version of the same. I have installed them for both POTS backup and as an interface to existing overhead paging systems with no problems.
I have also heard good reviews of the Audiocodes gateways. We have deployed several of their digital gateways, but have yet to use their analog FXO/FXS gateways.
Tim
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Hmm, looks like that file is not being used in the latest 3.0 version. So it might have been removed. Alternatives: 1) you can record a file and copy it as mb_to_copy.wav into the audio_en folder (you have to record with 8kHz, 16 bit mono or use the AA to record this file) 2) Upgrade to latest software version that is on the download page.
I have upgraded to the latest version and now I have encountered another problem with the voicemail box menus. When copying a message to another mailbox, I get the "message has been copied" recording, (and the copying works), then I hear the recording "message number" like it is going back to the messages, but the audio freezes at this point. The relevant log entries are listed below, after stripping out the .
[6] 2009/04/24 22:59:35: Received DTMF # [8] 2009/04/24 22:59:35: Play audio_en/mb_copied.wav [8] 2009/04/24 22:59:36: Packet authenticated by transport layer [6] 2009/04/24 22:59:37: Received DTMF # [8] 2009/04/24 22:59:37: Play audio_en/mb_number.wav [8] 2009/04/24 22:59:43: Packet authenticated by transport layer
Tim
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Have you tried downloading the latest audio files from the download page?
Yes, the file named in the log message is not included in the latest audio_en.zip (downloaded about 1.5 hours ago)
Tim
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Hello all,
I am getting an error in version 3.2.0.3143 that a recording is missing when I try to copy or move a voicemail to another recording. I tested this on a server we have that runs 3.0.1.3023 and I do not have this problem.
[8] 2009/04/20 17:13:53: Play audio_en/mb_to_send.wav audio_en/bi_7.wav audio_en/bi_9.wav audio_en/bi_9.wav audio_en/bi_6.wav audio_en/bi_press_1.wav space10 audio_en/mb_to_comment.wav space10 audio_en/mb_to_copy.wav space20 [3] 2009/04/20 17:14:01: Could not open WAV file audio_en/mb_to_copy.wav
On the phone I hear "To record a comment to the message" then a pause and it goes back to the voicemail message.
What do I need to do to get this resolved?
Tim
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Try to set a static registration in the registration tab of the extension at the bottom in contact field. Put the number and ip add the number and ip Address of the gateway like sip:1234@1.2.3.4
So the only way to do this would be to create phantom extensions for all the extensions on the other PBX? The pbxnsip server has extensions in the 7xxx and 8xxx range, the other pbx has extensions in the 4xxx and 3xxx ranges, and there are several hundred of them. Right now I have a trunk and a dial plan entry that routes the 3xxx and 4xxx over to the Audiocodes gateway.
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We have a client that has an old PBX they are keeping in place and a pbxnsip server for their roadwarriers. We have setup an Audiocodes Mediant 1000 with a T-1 trunk into the old pbx. Extension to extension calls between the 2 pbx's work without an issue.
Is it possible for them to dial into the AA and call an extension that is on the other side of the trunk with the M1k?
Tim
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Is there a * code or a way to pull a call back to your extension after it has been forked to a cell phone without hanging up and returning a call to the other party? I can see how this might be a potential problem with someone jacking your call if you are out of the office. Perhaps to help reduce that risk, there can be configurable option so people to have to enter their voicemail PIN to use this option.
Would anyone else be interested in such a feature?
Tim
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We have a client that we are configuring PBX for with release 3.2.0.3143 for their road warriors. The client wants to keep their legacy PBX in the office, so we are providing a Mediant M1k to provide a PRI to their legacy PBX for extension to extension calling.
I have configured the Mediant and the trunk on pbxnsip, and calling works both directions. The problem I ran into is I can't figure out how to make the pbxnsip trunk send the Extension # as the Caller ID number instead of the ANI. Does anyone know if there is a setting to make the extension be sent in place of the ANI?
Thanks,
Tim
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Oh I see! Looks like multiple "Entry" is not supported. We will fix it in the next release.
Thank you for confirming that, is there anything I can do to work around it for now?
While at it, I'd like to throw a feature request out there. From what I have been playing with, there are 4 other functions that would be very helpful. I know it is probably too late to try to get them built and tested for the next release, but maybe functions similar to these could be put on the roadmap.
CreateDomain - I am currently doing this with posts to the web interface, which was quite fun to get that multi-part form to work.
It would be much easier if I could create the domain within the SOAP calls as well.DeleteDomain - I currently have not implemented anything to do this, but it looks like it would have to be done on the web interface as well.
CreateAccount - This is not a current need for us, but the ability to create an account, so we can use the DBGet/DBSet functions to finish provisioning it would be very helpful.
DeleteAccount - Delete an account in a domain
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Tim
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Currently, it is designed to create 1 dial plan at a time. What you can do though, in your program, instead of sending the multiple dialplans at the same time, you can send one at a time in a loop.
I think I might not have been clear in what I am trying to create. I am trying to create 1 Dialplan with 4-6 Dialplan Entries. I tried calling CreateDialplan 2 times, first to create priority 101, then to create priority 102. After the second call, I get an error in the logs, "dialplan already exists".
[5] 2009/03/11 16:12:23: Receive SOAP request via HTTP interface <?xml version="1.0" standalone="yes"?> <env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://www.pbxnsip.com/soap/pbx"> <env:Body> <sns:CreateDialplan> <Domain>test5.aeoninc.net</Domain> <Name>Default Dialplan</Name> <Entry> <Preference>102</Preference> <Trunk>AEON Inc Outbound Trunk</Trunk> <Pattern>XXXXXXXXXX</Pattern> <Replacement>1*</Replacement> </Entry> </sns:CreateDialplan> </env:Body> </env:Envelope> [9] 2009/03/11 16:12:23: Other parameter preference, 102 [9] 2009/03/11 16:12:23: Other parameter trunk, AEON Inc Outbound Trunk [9] 2009/03/11 16:12:23: Other parameter pattern, XXXXXXXXXX [9] 2009/03/11 16:12:23: Other parameter replacement, 1* [5] 2009/03/11 16:12:23: dialplan already exists Default Dialplan
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Tim
It would be much easier if I could create the domain within the SOAP calls as well.
Replay custom message in Auto Attendant
in Auto Attendant Setup
Posted
I'm not sure what this has to do with entering an invalid option in an auto attendant, I am familiar with the timeout options and use them to time out to a general mailbox.
Tim