Vodia PBX
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That will break backward compatability with the product deployed in the field now. This is how currently it works, you allowed for domain to domain calling in V2 and now customers will not be able to do that as of V3.
Well, as said above in a server farm you cannot assume that a specific domain is physically on the same server. If that was possible in 2.0, then call it a bug.
Needless to say, it caused a lot of problems to have one call in two domains.
Can you answer the ANI question please?We have to document the ANI topic on the Wiki, answering that question here is the wrong place.
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Currently on a hosted server, extensions may call each other in different domains by using the TEL:alias. So domain2.com has a TEL:alias of 123456789, user in domains1.com dials 123456789, the call goes to the extension in domain2.com (it remains on-net and is classed as an internal call).
Does this security fix prevent this current functionality from working?
In a hosted environment, you cannot assume that a tel:alias is on a specific server. There is usually a server farm, and then you must resolve the name by using a trunk anyway. tel:-alias matching makes sense only for inbound calls that come from a global trunk.
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No, I am running a permanent key. Under status it says:
License Status: Appliance Key
License Duration: Permanent
The calls don't seem to be disconnecting when I am talking on them, just when they are muted which doesn't sound like a license limitation.
(A little bit at wits end)
Is there something like a firewall in between? Maybe the NAT closes after 3 minutes if there is no refresh from the other side?
We were able to reproduce the problem in the lab (though the timeout was not 3 minutes), and after the change it worked nice.
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How can we setup the Polycom sidecars to to start monitoring users just on the sidecar and not starting on the phone itself , there must be a way in the config file to do this , it just would look cleaner if all the monitoring started and finished in one place ..
You definitively have to switch to TCP or TLS transport layer (set in the PnP params). TLS requires a valid certificate in the PBX (http://wiki.pbxnsip.com/index.php/Getting_...lid_Certificate), if you don't have one better stay on TCP. APart from that, you should follow the procedure in http://wiki.pbxnsip.com/index.php/Polycom.
As far as I know there is no way of exactly saying what key gets what BLF. It is all dynamically assigned by the phone.
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I just checked incoming calls again. They re-route to my cell phone and voicemails work fine and get emailed correctly. Just no phone registrations.
This is on an SBS2003 server so the domain has not changed - I'm registering to an IP and not a name anyway.
I mean the domain name in the PBX, not in the server. Typically "localhost", see http://wiki.pbxnsip.com/index.php/Domain.
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netstat shows 5060 being used by pbxctrl.exe so I'm not sure what is happening. I can port ping 5060 to the server IP and that works. xlite and my phones still show 404 not found and why would the pbxnsip log files show the same? What can't it find?
I can receive calls so I know the service is working.
What about the domain name?
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Would a license assigned to a dongle work on MAC OS?
At the moment that is not possible. However, it seems that the dongle manufacturer supports OSX. So the answer is a clear "maybe".
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We are having trouble with our fax, so we switched trace on and we get 1000's of these messages in the trace. At first we thought it was the M3 phone, so we stopped that and these message still appeared.
What do they mean?
Here you can see how the DTMF detection works. The PBX looks at 9 frequencies (the one in the middle is the FAX detection tone). If anything is significantly standing out (2 is not significant), then that tone is treated as "present". A DTMF tone has one of the first four and one of the last four frequencies.
Needless to say, this kind of logging literally eats the CPU. So turn this only on if you want to debug something with inband DTMF.
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What is the ANI field for in the hunt group?
That's when the hunt group redirects a call to the outside world. Depending on the on the caller-ID presentation, you need that.
also, can you add a beep when you intercom/page someone? people have been complaining about not being alerted.IMHO that is clearly the job of the phone.
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I had a working system yesterday. Today when I went to check it all my extensions are unregistered and I'm getting 404 not found errors. I can verify that port 5060 is open, my polycom phones are downloading the config files from the tftp directory. XLite is also showing 404.
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Any idea what is going on? I have restarted the service and also the server with the same results. I was using the latest 2.0 version and swtiched to version 3. Replacing the 2.0 version back yields the same results.
You can check if there is another process using port 5060 by running netstat (same in Windows and Linux). Sometimes there are softphones which like to sit on that port.
If the port 5060 is really taken by the PBX, then check if you have a domain name or alias "localhost". Maybe you changed the name of the domain and then the PBX gets picky with the Request-URI.
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I recently updated my CS410 to the latest release 3.0.0.2992 hoping to solve a one-way audio issue with my snom3xx phones. Now as each of the snom3xx phones rebood that enter a loop "Sending DHCP requests..." DHCP is disabled on the CS410 and DHCP is supplied by the network router. I guess that solved the one-way audio problems for now by basically turning the phones into paper weights.
If I press the settings button on the phone it will boot up into the phone application without an IP address. Then when I clear the settings the phone reboots and gets an IP number. Then, it looks like the phone recognises new software and reboots again. This time it will re-enter the DHCP loop.
I would run Wireshark in the network to figure out what is going on there. If you have port mirroring on your switch considering using that to see what the phone specifically is doing.
In general, having two DHCP servers in the network should not cause such a problem - this would cause different problems.
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My e-mail service uses port 26 for smtp. In the e-mail configuration for my cs410 have the address configured in the form mail.domain.com:26. When I initially received my cs410 this worked. I can't remember what version of software I updated to, but at some point this stopped working and in the logfile I have a repeating message "SMTP: Cannot resolve mail.domain.com:26"
It appears that the cs410 with version 3.0.0.2992 no longer can resolve the port re-direction. I have turned off any e-mail notifications to prevent the memory space filling up with messages pending to be sent.
Seems that there was a bug creeping in. Try http://pbxnsip.com/cs410/update-3.0.0.2995.tgz.
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Does anyone know where I can look for information that will allow me to authenticate on a Snom370 or similar. I had a number of problems with my Router and the Snom and because I reset it to Factory Default, I am having to reprogram it. Not so difficult, but I have just gotten the unit.
Is the authentication required of the telephone with the Extention Registration in pbxnsip?
That sounds to me like the password in the PBX differs from the the password on the phone. Try re-setting the passwords.
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I have some agent groups (my company departments). For example I have AG with 4 static members. I've detected very strange behaviour of this and other AG, when caller have no Caller-ID we have ringing 1 or 2 of 4 phones, but when there is normal Called-ID all phones are ringing. What's the source of this problem?
For info: all phones are Linksys SPA-921 and Linksys SPA-941.
That is indeed very strange. Is that reproducable? Do the phones reject the call? Maybe there is something that makes them reject anonymous calls.
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Do the Message Summary lines under the registration tab of an extension show as Registrations for SNMP Reporting?
message-summary 901 sip:901@192.168.0.2:2051;line=1wvo15xt 59
Yes, they also count. I think that is reasonable, because the SNMP stuff is about resource usage of the system, and a MWI subscription is as expensive as a regular extension registration to the PBX.
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The problem with not recording 0 is that then when someone does hit the queue because the agents are busy, then callers just hear music with no explination of what is going on with their call for the first 20 seconds or so. If you decrease the timer, then they are constantly hearing the recordings, and not the background music.
Well the first time someone enters the queue should be only half of the waiting time. But I get the point. Maybe we should extend the syntax for the waiting time and say the first number if for all prompts and the second (if present) specially for the first prompt. Default would still be the duration divided by 2.
Gap between announcements (s): 20 5
Would mean: Usually wait 20 seconds, and for the first prompt wait 5 seconds. What do you think?
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Version: 3.0.0.2899 (Linux)
In any case I would suggest to update to http://www.pbxnsip.com/cs410/update-2992.tgz. Maybe it already fixes the problem. Make a backup before doing that, just in case anything goes wrong.
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I have installed the 3.0.0.2994 version and it looks like my language settings are not working anymore for the snom phones.
I had placed the files Snom_gui_lang.xml and Snom_web_lang.xml into the html directory and the gui_lang_NL.xml together with the web_lang_NL.xml files into the html\snom subdirectory. After resetting the snom phones they now are comming back with the english language.
Two things... The first thing is that obviously something was broken and the language provisioning for snom wasn't working out of the box (that should be fixed in the next build).
The other thing is that the automatic provisioning works only for languages that are available as web interface languages. That might be insufficient. Maybe we have to introduce a PnP parameter for this (though I really like the idea of controlling that through the vendor-independent language settings for the account).
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I don't think I am using the demo key. I should have a licensed CS410 and did before. Upgrading the software wouldn't have affected my license would it? How would I know if my license is limited to 3 minutes?
The name of the key would be something like "3 Minute Demo". That string shows up under admin/status.
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HI;
i am an it consultant who has setup a number of PBXnsip phone servers over the last few months.
i am enthralled with the new features version three has to offer especially now that it has been released for windows servers too!
however i am having a major issue and my clients are going live tomorrow morning with no other phone system!
when i go into the auto attendant to upload files it does not not always add the file to the recording folder in the pbx system folder.
secondly the ones that did upload now give the caller an earful of noisy static instead of the pleasant voice of the auto attendant talent!!!!
If you have access to the file system, there is a simple workaround. Just record the prompt with on of the *98 codes and check what file has been written to the recordings directory. Than replace that file with the recording that you actually want to load through the web interface. That should save your day.
For the static noise, make sure the file is 8 kHz sampling frequency mono, 16 bit/s sample WAV format.
In the meantime we'll check what going on with the upload.
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We have tried everything.
The problem ONLY shows on ext. there are in an agent group.
The PW are very secure - and, nope, no one is teasing.
The second the ext. are moved from the agent group into a hunt group, the problem no longer excist.
Okay, what version are you on the PBX? Where exactly do you see that the extension is on DND in the web interface? Did you ever hit the save button on the DND web page?
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I do not see how this will help, as I have two phones standing, not beeing used, and just by looking via the web-interface I can see they change status.....
You see the DND status in the web interface changing without someone touching the phone? Do they have a good password set for the web interface? Maybe someone has some fun teasing you? Or can you trace the SIP traffic to the registered phones? Maybe they get bored and start calling the PBX when they are idling...
There is no little genie in the PBX that does things like that...
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What to do - what can I have done wrong????
Maybe there is a speed dial somewhere for DND.
I would turn sending emails for status change on, then the user will get an email when the status changes. The user will tell you what he did in order to get this email ("I did nothing, just press this button").
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Hi I have been trying hard.... to get the Conference information to be sent out to participant.... but I'm still not able to do it.
Basically, the rest of the email alerts works e.g. CDR call records, miss call alerts and etc.
However, whenever I try to create a scheduled conference, the email is not sent to my participants.
I have specifically followed the instructions by keying in either ext no. or emails.... both doesnt work.
Please advice...I'm currently using the latest PBXNSIP 2.1.14
Does the moderator have a email address setup in the account? Only the moderator should receive an email, not the participants. It should also work for adhoc-conferences.
The following participants are currently in the conference:
StartNumber2008/08/17 09:01:17482008/08/17 09:03:2144
Do not reply to this Email. It was sent automatically.
Outbound Caller ID
in Calling Card Setup
Posted
Okay, that would be a bug then. Needs to be fixed.