Pradeep
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Posts posted by Pradeep
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Would be glad to. Should I just delete the following directories:
/Library/pbxnsip
/Library/StartupItems/PBX
/private/var/run/pbx
Is there anything else to remove? I ran into this error the first time I installed the software (which was with the latest installer).
No, that should do it.
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Seems like there is some issue with the post install script. I will take a look at it. Meanwhile, could you please make a backup of your pbx working directory (you can find the working directory location by going to Admin->Status page), remove the old installation manually and then reinstall the desired version?.
I'm using an Intel-based MacBook Pro on 10.5.5 and am getting the "The following install step failed: run postinstall script for pbx-mac. Contact the software manufacturer for assistance."Here's my log:
Nov 19 20:08:18 MacBook /System/Library/CoreServices/Installer.app/Contents/MacOS/Installer[12003]: vm_allocate: 0, 0x5800000 - 0x25800000
Nov 19 20:08:18 MacBook /System/Library/CoreServices/Installer.app/Contents/MacOS/Installer[12003]: vm_protect: 0
Nov 19 20:08:18 MacBook Installer[12003]: @(#)PROGRAM:Install PROJECT:Install-384.1
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So it's best to wait for 3.1 to be released, before working on the address book.
When you think this will happen?
I understand.. we were just studying the manual, and the great part of existing examples are US oriented.
Anyway, right now we have 2 outbound trunks, one is the Gateway and the other is the VoIP provider.
Due to local rates, we are using mainly the PSTN and the VoIP only when we have no more PSTN lines availables.
To select the line, we just type '8' or '9' in front of the number.. any tip on this?
Can we tell the pbx to use the VoIP trunk when the Gateway is full?
Thank you for your answers.
Nicola
3.1 is ready for the testing. Please take a look at the posting http://forum.pbxnsip.com/index.php?showtopic=1650
You can create trunks and dialplan entries to achieve this.
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i was curious as to how remote call control was going with pbxnsip? I heard that CSTA is in the works.
Thanks
Joso
Currently we do have basic support for the CSTA. Please check out http://wiki.pbxnsip.com/index.php/CSTA
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manually running ./pbxctrl-darwin9.0-3.0.1.3023
results in this
-bash: ./pbxctrl-darwin9.0-3.0.1.3023: Bad CPU type in executable
so I assumenow during compilation G4's or dual G4 was forgotten.
It is not forgotten. Currently, we support only Intel based CPUs
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I am attaching the log of the tracert call to analize
Can you try changing the "Ringback" trunk setting (at the bottom of the trunk edit page) and see that changes the behavior that you are looking for? May be the VoIP provider is ignoring the '183 Media'.
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in the pbx.xml?
Yes, in pbx.xml
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We have a simple dial plan "9*". So to call to the outside 91xxx-xxx-xxxx.
However, when we receive a call it shows up on our CallerID on the Cisco phone as 1xxx-xxx-xxxx.
As a result when we use redial it does not work because the number does not have 9 prepended to the number.
What would be the best way to fix out problem?
We are using pbxnsip 2.1
Thanks
You can add another dial plan 1xxx-xxx-xxxx to 91xxx-xxx-xxxx
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Rightnow this field is still empty. I guess this field needs to be filled with a extension.
Can you give me some help?
Please take a look at Microsoft Exchage UM http://wiki.pbxnsip.com/index.php/Microsoft_Exchange. It talks about the field that you are interested in.
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The Comtrol UHLL protocol is a great addition. However a great deal of the systems for call logging, property or hotel management require serial feeds rather than an IP feed which is of course more elegant than serial. Can you not offer both serial and ip connectivity?
What kind of call logging are you thinking? I am not sure what is missing by having IP interface over the serial feed.
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Generally, if you can prefix '8' in front of the extension, then it sends directly to the voicemail box of the extension
If the call has to go to the mailbox always, you can set the "Call Forward all calls to" 8<ext>, where <ext> is the extension number. This setting is available under the "Redirection" tab for the extension.
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I'm trying to make an AA which will transfer the call directly to the voicemailbox to an extension (which is normally available).
The extension is normal available. (not busy, not unavailable)
But when this AA is called the call has to be redirected to (no delay) the mailbox of this extension.
How can I achieve this?
Generally, if you can prefix '8' in front of the extension, then it sends directly to the voicemail box of the extension
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hey,
yesterday my cs410 quit sending emails, but it sent a CDR, now I cannot get it to answer calls or call out on it, it will answer an incomming call on the fxo ports, but just goes silent..I can ping it and SSH into it, but the webserver no longer will run...any ideas on what happened?
Could you please tell us what is the software version? Since you can ssh, can you check the available disk space?
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we are getting these details from pbxnsip log
[5] 2008/10/07 06:39:30: Identify trunk (line match) 3
[5] 2008/10/07 06:39:30: Trunk calllcentric sends call to 17772335784
[5] 2008/10/07 06:39:30: Trunk call: Could not identify user
wat does dis mean and wat shld we need to continue
Could you please send the call log starting from INVITE? That way we can see the complete details. BTW, you can increase the logging by using Admin->Settings->Logging page and setting "Log Other Messages" to "Yes" and increase the log level to 7 or above
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Some time ago iv'e seen a topic like this but can't find it anymore, but I want to call with an extension from one domain to another I tried to set the trunks to "global" but this didn't take effect, how do I need to configure this?
You can set the "Accept Redirect" on the trunk and call should go through
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yes we chked it. but we are not getting which number we shld call from x-lite so dat it goes to our speech server application through pbxnsip.
From the pbxnsip side all you need is to create a trunk that points to the speech server address, dialplan that uses this trunk and assign the dialplan to the account(x-lite extension). On how to create the trunk and the dial plan check out these links
http://wiki.pbxnsip.com/index.php/Trunk_Settings
http://wiki.pbxnsip.com/index.php/Dial_Plan
After the configuration, dial the number that you specified on dialplan and that should send the call to speech server.
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Hi
we have already a speech server 2007 application. we hv got a 3 minutes license frm pbxnsip.
we have created an account in http://www.callcentric.com/ and added this as a trunk in pbxnsip adminstration interface.
we hv also created an 9999 extension. we hv configured the http://www.callcentric.com/ account on x-lite also.
pls guide us how to integrate between pbxnsip, our speech server application and x-lite softphone.
Regards,
Shwetha R.K
have you checked out the forum entries in section http://forum.pbxnsip.com/index.php?showforum=62 ?
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Yes I did.
Make sure that you have copied the 'Win32 update' not the installer.
You can also, try to run the new installer first and the perform the manual upgrade process.
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where is the ANI field located in OCS or PBXNSIP. We don't use a gateway and connect the OCS Mediation server directly to PBXNSIP?
ANI field is available both on the trunk and the extension level. (Note: on the trunk page look for "Trunk DID", if you are using the official 3.1 version)
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I installed it on my 2008 server but it doesn't show in the Phone and Modem options.
I have not worked on Windows 2008 server. But I assume that it has the "Search" option when you click on the "windows/start" button on the bottom left corner. You can type in "Phone and Modem Options" and windows will search it for you.
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Hi:
I am trying of to record a current conversation using the record button of one Snom 320; I have configured the email address at extension of the Snom 320, and I m using a license with call recording, also I have used a version without call recording.
I am expecting a mail to such email address attached with the wav produced by the recording, but it does not happen, all the other messages like voice mails arrive to the email address, but file with recording does not .
I renember some previous version where this feature was working fine, by now I use 3.0.0.2998.
Any body can tell me what is happening?
Thanks and best regards.
Juan Acevedo
Can you please verify the "recordings" folder under the pbx install directory? If the file is generated there, then there must be some issues in sending emails.
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Hello,
I'm trying to get the TAPI driver working.
When I try to call using my SNOM360 th PBX log shows:
[7] 2008/10/02 10:04:11: SIP Rx udp:127.0.0.1:1034:
NOTIFY sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:1034;branch=z9hG4bK-r433tyk1h8osuw4fjcwz;rport
From: <sip:32@hofstee>;tag=23487
To: <sip:32@hofstee>
Call-ID: eilt76f47xpey71wsqio
CSeq: 2343 NOTIFY
Max-Forwards: 70
Contact: <sip:32@127.0.0.1:1034>
Event: x-tapi
Content-Type: application/x-tapi
Content-Length: 59
Action: MakeCall
Line: 1
Call: 250
Address: Phone number to call
[7] 2008/10/02 10:04:11: SIP Tx udp:127.0.0.1:1034:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 127.0.0.1:1034;branch=z9hG4bK-r433tyk1h8osuw4fjcwz;rport=1034
From: <sip:32@hofstee>;tag=23487
To: <sip:32@hofstee>
Call-ID: eilt76f47xpey71wsqio
CSeq: 2343 NOTIFY
Content-Length: 0
What could be the problem? the snom is not rinning and the Snom sip-trace log doesn't show the call.
Looks like PBX received the NOTIFY without the address filled in.
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Is there a RedHat version available? We've been testing 3.0.1.3018 Linux.
Thanks.
Here it is www.pbxnsip.com/download/pbxctrl-rhes4-3.0.1.3023
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Logfile
Clear or Reload the log.
[4] 2008/09/30 16:12:05: Translation item email_queue.htm#dur_talk#fr not found
[4] 2008/09/30 16:12:05: Translation item email_queue.htm#duration#fr not found
This is harmless. Just some debug info, will be removed soon.
Installation fails with run post install script with pbx_mac
in MacOS related topics
Posted
Working on it. We had some upgrade issue sometime back. I guess fixing that caused this issue.