Vodia PBX
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Thanks for the files. now the next step how will i install them? i put them into PBX folder, i restarted the system but still in the settings, i can not find a choice for turkish.
You mean that you see a directory audio_tr with the WAV files in it, but you cannot see the option in the web interface? WHat version are you running?
Last resort is to rename it to audio_sp and select "Spanish"...
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We have a prospect that is interested in running pbxnsip on a blade server with many instances of pbxnsip bound to different cores and across 10 blades. There could be 500 domains if all goes well. They are asking for a tool to manage all the domains instead of http'ing into each system to manage the individual domains on that system. Any ideas on how to do this without to much effort?
Sooner or later they have to write some shell scripts that automate their processes. curl can be of great help, as most of the day to day steps can be done through the web interface. But other jobs like managing DNS obviously needs to be done outside of the PBX itself.
The big question is if they want to give the customers access to the self-adminstration web interface or they want to do this through a set of scripts that use SOAP to change certain settings on the PBX.
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When a call is transfered from one extension to another, the caller hears a ringing sound.
Is it possible for the caller to hear the Music On Hold sound instead?
You mean on blind transfer? It is a kind of feature, because it tell the caller that the other side is now ringing and someone is (hopefully) running to the phone. But it is an interesting point - the caller does not really care if it is a blind transfer or attended transfer. Maybe we can make this a setting. I believe the current behavior is fine as default behavior, but maybe someone wants to play music instead.
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What would the Invalid incoming gateway SDP be. I have not set any preferred codec in the trunk settings.
The SDP offer is in the INVITE packet. It should be just "above" the trace you showed.
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I did the update to 3143.
Afterwards the incoming calls worked but outgoing calls did not. Here is what I received in the log when trying to call out.
INVITE Response 404 Not Found: Terminate 887c3383@pbx
It would change the number in front of @pbx of course. I checked all the dial plans and such. Trunks looked up and working.
I didn't work on it all too much as I had some incoming DID lines not working as well. I spent a lot of time on the incoming issue first and come to find out a switch was down due to a patch cable being used by someone there on site. He needed a cable and borrowed one and it was the same time I was troubleshooting for 45 minutes.
My method was I dumped the configuration files. Upgraded the PBX and restored the config files. I did noticed I now had duplicate trunks listed under dial plans when selecting a trunk in the drop down. Only one trunk was visible under trunk settings. Well besides the default pstn trunk.
Should I have not restored settings? I also noticed it handles the alias number a bit different. It adds the number to the back of the account. I only have two DID lines and they both worked after I solved the missing patch cable mystery.
Side note. On the lan side i found the router was port forwarding all ports from 20,000-60,000 as RDP. I changed this to port 3389 for now. This is lan side only and the 410 has no lan gateway. The pbx has its own static IP. Just wondering if this could have caused the one way audio issue. I also updated that router wgr614 v9 to the latest firmware while onsite.
I will attempt another upgrade after I hear back and can schedule more downtime.
Usually the problem is that the 3.2 version tries to be smart and interpret numbers if you set the country code. One question is how many digits you are using for national calls (10/11 digits). Then in the dial plan, in USA you must use 10 digits always. If you use the old pattern 1* you probably run into problems.
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It's still doesn't work as well. The same - instead of custom zone name from timezones.xml I see in the blank line.
About additional fixes for RU:
1. This version include few new timezones (Samara, Ekaterinburg, etc). Fisrt of all - this zones named in English. Strange to see it when whole interface translated to Russian.
2. Small mistake: change please Yakrutks to Yakutsk. It's typo.
If you are using the 3160 build, then just move the timezones.xml files out of the way.
We added the Russian translations for the time zones, should be looking better in the next build!
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Any having a working download link????
Currently we are at http://www.pbxnsip.com/protect/pbxctrl-3.3.0.3160.exe
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i cannot find that option in 3160 and there is not reference on the wiki.
Yes, that option is obsolete now. Now you just need to trust the MAC on the cable, this is now in the PnP tab.
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I wonder why pbxnsip can't recognize and announce T38...
T.38 is "not trivial"... If the PBX would have to advertize it after detecting the CNG signal, it would be responsible for converting G.711 into T.38
. I totally agree, the whole situation is not very satisfactory. I believe almost the whole VoIP industry agrees... Modems were designed for analog lines, not for digital RTP packets.
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i am registering an Asterisk PBX to the phone system and it is registering just fine however the users on the Asterisk PBX are complaining of alot of static and calls dropping all the time.
are there any additional steps i should take to get this fixed? i have regular phones registered using the same trunk that are having no issues at all.
Static and calls dropping are usually a sign of network problems. I guess the PBX is not in the same LAN? Maybe the connection is a little bit instable.
Can you see who is sending the BYE message? Do you see a Reason header that explains why the call was disconnected?
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I have pbxctrl-rhes4-3.3.0.3147 is there a newer version? If so, possible link??
http://pbxnsip.com/protect/pbxctrl-rhes4-3.3.0.3160 is the latest & greatest.
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Thank you. Is this release going to be GA soon?
Fingers crossed... I believe so!
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Translation file on the same folder. Everything is ok.
OS: Windows XP
Tryed on 3.1 and on 3.2.0.314
If you like, please try http://pbxnsip.com/protect/pbxctrl-3.3.0.3160.exe. That might also contain additional fixes for RU.
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Since we upgraded PBXnSIP to ver 3, we cannot recieve calls to the Communicator clients anymore. Is there a document available for the integration for ver 3.x as the one available is for ver 2.
We did not change anything else, yet incoming calls to Communicator from PBXnSIp no longer works.
Are you using a country code? That might be a problem; and you can also check on how the PBX represents global numbers on the trunk. OCS likes the "+" notation, at least before R2.
A look at the SIP trace should help finding the problem.
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I may have something mis-configured but I'm having an issue recording calls on version 3.2.0.3143 (Linux). Recording defaults for the domain are set to no. On the account that requires recording, I have the recording settings "Record outgoing calls to external numbers on", "Record outgoing calls to internal numbers on". Destination for recordings is "recorded-calls/$m/$d/$u-$t-$n.wav". However, recording did not work with the default location either. This worked on a previous version without fail. I can't seem to get it to work on this version with the settings mentioned. Linux is RedHat EL5.
Using *12 and attaching recording to email does work.
Yea, that's a bug we already fixed in 3.3.
http://wiki.pbxnsip.com/index.php/Release_...#Call_Recording
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Where is that setting on the snom M3 I can't seem to find it. Also we are getting terrible feedback with the speaker on these M3s. I know it is the speaker feedbacking into the MIC and this causes the speaker to cut out. Any ideas?
On the M3,under "Management settings", set the "Configuration Address" to the IPv4-Address of the PBX.
If you use a later version of PBX 3.3, then you will even get the right time zone
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Is it possible to set a timer on the cell phone voicemail notification? I would like to be able to do it on a per extension basis. My main goal would be to shorten it up so it gives up prior to leaving a message in my cell phone's voicemail.
There is a global settings called cellphone_timeout. The default is 20 seconds. You can change it in the Global Configuration File, see http://wiki.pbxnsip.com/index.php/Global_Configuration_File.
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What's wrong? Advice pease!
The translation is missing? Any change to try an upgrade? If so, what operating system?
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Have we overlooked a common feature in voicemail systems - Volume Control when listening to vm from remote access?
This topic pops up from time to time. The PBX does not change the volume. We could change the volume during the recording, that would not be such a biggie. But IMHO having problems with the volume is a clear sign that there is something wrong with the gain of connected devices and it should be corrected there.
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In version 3.3, we are doing some neccessary changes. The good old MAC-address based config only works if the phones are in the LAN and the server can see the MAC address on the cable (layer 2). Otherwise, you have to provide the username/password in the phone (see http://wiki.pbxnsip.com/index.php/Snom#Set...name.2FPassword). Probably that hanging is because of this.
Also, consider using the latest 3.3.0.3160 version, http://pbxnsip.com/protect/pbxctrl-3.3.0.3160.exe.
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This post is as clear to me as the XML config from Cisco is...
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[5] 20090311120259: Dialplan voip: Match 2223334444@domain.com to <sip:12223334444@callcentric.com;user=phone> on trunk callcentric-domain
Sounds like a problem with the 10/11-digit representation. If you have country code == "1" then use only 10-digit numbers in the dial plan.
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i want to set the following parameters which are not in the PnP settings in the admin interface.
holding_reminder!: off
user_ringer1!: Ringer7
ring_sound!: Ringer7
alert_internal_ring_sound!: Ringer7
alert_external_ring_sound!: Ringer7
i looked at this post but cannot figure out how to get this work
http://forum.pbxnsip.com/index.php?showtop...8&hl=ringer
any help? thanx
Put the following file into the html directory:
<?xml version="1.0" encoding="utf-8"?>
<phone-settings>
<holding_reminder perm="RW">off</holding_reminder>
<user_ringer1 perm="RW">Ringer7</user_ringer1>
<ring_sound perm="RW">Ringer7</ring_sound>
<alert_internal_ring_sound perm="RW">Ringer7</alert_internal_ring_sound>
<alert_external_ring_sound perm="RW">Ringer7</alert_external_ring_sound>
</phone-settings>
Check the log about what file is missing in the provisioning process. Could be "snom_320_custom.xml" or something like that.
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Pbxnsip 3.3.0.3156 snom 320 Auto provisioning not working
in snom Phones
Posted
Oh, are you using MAC? Grrrrr. MAC does not have /proc/net/arp... The PBX now needs to find out what MAC is what IP address. That little part is missing in the BSD implementation yet!