Vodia PBX
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i tried but result still same. remain showing pbxnsip version 3 logo.
Try clearing the browsers cache, or just F5.
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i tried it and it took the entry okay but the outbound ani is still the xxxx primary ext number
Try changing the RFC3325 mode on the trunk. Depending on the mode (gateway/register) and the presentation you may get different results.
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after doing an .exe only upg to 2.1.9.2467 we lost our customization/logos. the system settings under apperance also reverted to www.pbxnsip.com, etcetera. easy enough to change back but need to know if this has happened to others or if something did not go right when 2.1.9.2467 ran?
What was the previous version?
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when you hit the transfer button and enter the ext and then press transfer the call gets dropped.
Whow. Do you have the SIP messages in the log for that?
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after upgrading to 2.1.9.2467 we noticed that some (random) phones can not register any longer. scenario is customer calls stating their snom phone states NR on the lcd and when we check the sip logs we see "authentication required". after changing the password and talking the end user through entering the new password they can register again. i am wondering if this has something to do with the stronger password requirements i am seeing with 2.1.9.2467? about 20% of users are having this problem.
No the password policy has nothing to do with that (it only happens as JavaScript in the web browser).
Hard to say what went wrong. Maybe the snom wizard kicked in and entered a new password.
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many of our users do not like the default vm greeting playing their 10 digit did number for their mailbox. so we setup their ext with the 10 digit d.i.d. in the alias and the four digit ext number as the prime ext number. this is fine until they make an outgoing call and the ani goes out as a four digit xxxx ani. i know we can block the ani and that users can record greetings to deal with the greeting issue in the first place but i do agree with users that in some cases the ext# as prime works better. so i need to know if their is anyway to make this happen?
If the alias list contains a tel:alias, the PBX will use it as ANI for outbound calls. For example:
Primary: 4566
Alias: tel:2121234567
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I am right in saying that after changing the registry with that dword entry pbxnsip as an application will now have access to changing the packets through winsock, I though it would have been more complicated than that. Should I leave the default value in the global config as 184 (0xB8) or should it be higher?
Yes. 0xB8 is usually a good choice, that is why it is default.
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The phone always expects GMT, and it adjusts it based upon the timezone specified in the xml config file that the phone downloads from the pbx at bootup.
Okay, we put that into the head version, will be in the upcoming releases of 3.x.
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Is there any way to define a * code or something like that. The problem is that if we define a AA as an MB Escape and users over-ride the domain setting as many of them do then this is not consistent. We would like to be able to tell everyone to hit xxx to go to another mailbox. Oour old system was *T or *7 for transfer.
Well, they can hit *0, the star is just being ignored though.
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I'm concerned about the time stamp when you listen to messages. The voice recording saying the time right before you listen to the message so you know when they called. Looks like pressing 5 gives you the info. Any ideas on being able to rewind while still in playback if you miss something?
The rewind only works on the message itself... When you are listening to the envelope, there is no stop or rewind.
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The default for the Voicemail duration field under the domain settings appears to be blank. How long of a message can someone leave if you keep this field blank?
If there is nothing set, the duration is limited to the maximum duration in the system (reg_settings.htm), usually ten minutes.
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here is logo link path entry = http://127.0.0.1/img/main_logo_blank.gif
Oh, in 3.0 you must put the relative path there (img/main_logo_blank.gif), not the absolsute path.
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Could someone please tell me how to implement QoS for pbxnsip in Windows Server 2003.
In Windows, you need to change the registry to enable the tagging of RTP packets for the Type of Service. There is a article from Microsoft on this topic: http://support.microsoft.com/kb/248611/en-us (see http://wiki.pbxnsip.com/index.php/Installing_in_Windows).
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I have discovered through Cisco's documentation that the 79x1 phones first tries to set the date and time off of the SIP registration responce, and then they fine tune it off of the NTP server that is listed in their config file. I am having an issue that since the date, and time is not sent in the SIP messaging to the phone, that the clock never gets close enough to use the NTP server. Is there a way to make PBXnSIP send the date, or does anyone know of a workaround? Below is a copy of how a Cisco sends the date from their callmanager.
Whow. Actually it is no big problem to put that in. Is GMT always fine or does the phone expect "local" time (wherever the phone is)?
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Well the phone thinks that this is DoS. It wants you to send the request from the PBX IP address and port. There is a option to turn this off on the phone, but then the phone gets vulnerable for simple DoS.
Same thing for TCP. You can set the phone to accept TCP connections (which is off by default), but then again you risk that someone plays DoS with the phone in your network.
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mmm...yep. The outbound proxy setting is empty in the trunk. Because of that the local phones works perfectly but not a remote phone? what is the correct input for the outbound proxy setting for the remote phone to work?
When a SIP INVITE comes in, the PBX needs to make a decision if that call comes from a trunk or an extension. The outbound proxy of the trunk is used for that - if it is missing then it is wide open and matches any incoming request. The PBX later assigns the call to an extension if the authentication is okay, but until then it is treated like a trunk call.
So it is easy, just set the outbound proxy and give it a try.
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Does anyone have a VOIP Client recommendation for Windows Mobile devices with WMD 5/6. ... and have it working correctly?
Counterpath had a softphone, I saw it running on a Samsung cell phone.
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Check if you have a trunk without outbound proxy. That can make the PBX believe that the call comes in on a trunk, and not from the phone.
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I have recently upgrade from version 1.5.6 and we used to have a time and date stamp that was attached to every voice mail. Since upgrading to version 2.1.8 we have lost this "feature". I have looked through the doc and the wiki and can't find anything on the subject. Does anyone know how to enable this "feature"?
Yea, the timestamp is the timestamp of the email. Even if there are a few seconds delay between the recording and the email time stamp, sobody seems to have a problem with it.
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Does it work if i setup a hunt group in pbxnsip and add extensions of some OC-Clients?
The goal is that our main phone number rings on multiple Office Communicator Clients
Well, the communication between OCS and the PBX currently runs through the mediation server, which is like a PSTN gateway (but translates OCS SIP to plain SIP). If you want to ring communicators that are registered on OCS - that sounds difficult to me. It would probably be much easier to register the communicator clients on the PBX. But I am not the expert here, check out http://wiki.pbxnsip.com/index.php/Office_C...ications_Server.
Gateway: If they are talking SIP I am pretty sure it will work. PSTN gateways are easy to get working.
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But I can not call external telephone numbers - only the local numbers to the office.
That sounds like a problem with the assignment of a dial plan to the extension. The domain has a default dial plan which applies to all extensions, check if it is set. Alternatively, you can also check the dial plan that has been assigned specifically for that extension.
If you can call other extensions, that means that there is no problem with NAT.
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How about the ports in the above question?
One port is probably for the PBX and the other one for the PHP. You can find out what ports are being used by which application with the netstat command.
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If you can afford to restart the PBX you can also change the files on the file system. Then you can avoid the whole SOAP trouble.
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[9] 2008/05/16 15:56:45: DTMF: Power: 0 0 0 0 0 0 0 0 0
[9] 2008/05/16 15:56:45: Last message repeated 2 times
[9] 2008/05/16 15:56:45: DTMF: Power: 0 0 2 0 0 0 0 0 1
[9] 2008/05/16 15:56:45: DTMF: Power: 0 0 0 0 2 0 0 0 0
[9] 2008/05/16 15:56:45: DTMF: Power: 2 0 2 0 0 0 0 0 0
[9] 2008/05/16 15:56:45: DTMF: Power: 2 0 0 0 0 0 0 0 0
[9] 2008/05/16 15:56:46: DTMF: Power: 0 0 2 0 0 0 0 1 0
Well obviously there is inband DTMF detection going on, but it seems the volume is too low to have a reliable DTMF detection ("Whisper DTMF"). Maybe whoever is sending the DTMF should increase the volume.
SIP Session Timers?
in General Setup
Posted
Well, because the PBX is media-aware session timer are not neccessary. What happens if the call gets esblished? Do they reject the call?