Vodia PBX
-
Posts
11,130 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Vodia PBX
-
-
Recently I noticed that I am not receiving VM in to my email box. After some checking in lto ogs I realize that there is a typo for our mail server:
[3] 2009/01/29 22:00:10: SMTP: Cannot resolve mail.rockbridge.ne6t (should be mail.rockbridge.net)
I looke everywhere and cannot find it. Every screen appears to be correct. Is there a file on the server which holds this information so I acan manualy change it?
The SMTP settings are either in the "Logging" settings on admin (really difficult to find, but we already moved it into a own email section) or in the domain settings.
-
I thought the 11-digit would fix it, but it made no difference...
As for the dialplan, I had the country code set to 1. The users were dialing 1 (then AC & local number). Still got the message. I guess I could try adding the 1 (1*), but if they're already dialing it, and the dialplan is * * shouldn't it get through?
It is strange. Maybe we can "catch" the situation also here in our test system.
-
is there a way to tell the pbx to flush that cache. this just started to happen since callcentric did some work on their network.
no changes on the pbx.
after a restart it works again for some time.
The PBX is just obeying oders regarding the TTL. If the service providers thinks that it is enough to check once per day if a record changes it is a "feature" that the user has to wait one day until the DNS record changes.
Though I have some doubts that that was the problem. I know from other users that they did not have such a problem with callcentric.
And you can always override the DNS by manually looking it up and putting the IP there temporarily.
-
hey, it works now...dunno what i did.
Well, there might have been an upgrade problem. Previously the settings could be "true" or "false". In 3.2, domain admins can specify in much more detail who is allowed to do those interesting things. "true" wound now be "*" (everybody), "false" would be "" (nobody). And now you can just list the extensions that have permission.
-
FWIW downgrading to 3.0.1.3023 fixed the problem.
I'd still like to know what the deal-e-o is so I could be running the newer software, but for now all is well...
Oh, in the 3.2 version the trunk actually has a setting for this. The "11-digit" representation is probably what you are looking for. You can also solve the problem in the dialplan with a replacement of "1*" (if you have no country code set on the domain level).
-
we have over 10 domains
Okay, then things are not so easy. Check out "How the PBX identifies the trunk" in http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk. There was another post today where the topic of multiple domains was covered (see http://forum.pbxnsip.com/index.php?showtopic=2065).
-
Thanks for the help!
Next version will have a big warning sign on the web page if you have only one domain but not the name "localhost" in it...
-
We have multipel domains, and for every domain i built the same set of trunks,
how exactly does the GLOBAL thingy work? can i build the 5 trunks were using and let all domains use it? what problems am i running into? and what do i need to avoid? and better yet, how do i go about setting it up?
we use approx 5 trunks for outgoing, and 1 for incoming traffic
There are several ways to do it.
1. You can just have one global trunk for all domains and share it amongst the different domains. For this, you would typically create the domain "localhost" with that trunk in it and customer domains with their respective DNS names. Inbound traffic will be routed based on the DID number and sent to the right domain. If you select the country code (e.g. "1") then the PBX will make sure that the 10 or 11-digit represenation is not a problem. All domains can see this trunk in their dial plan; however are not able to change it.
2. If you want that each domain may have their own trunk, then you should try to use a registration trunk. The service provider must fully support RFC3261-URI, especially including parameters like the "line" parameter, so that the PBX can match inbound traffic easily and safely to this trunk. This approach has the benefit, that the customers can use the CO-line features of the trunk.
You may also mix choice 1 and choice 2. For example, for inbound traffic you can use 1. and for outbound traffic 2. If you have only outbound traffic on a trunk, then make sure that you tell the PBX so in the trunk direction dropdown.
Number 1 will make your life a lot easier if you are using a regular PSTN gateway with just one IP address. For those kind of setup's it is otherwise quiet difficult for the PBX to detect where the call should actually go to.
Number 2 can also be done with gateway trunks, but then the trunk matching will become tricky. In this case you have to be strict on the DID number definition.
-
it looks like 7.3.14 / 7.1.39 are out.. does this fix it? will test later.
Yes, that should include the dial tone on shared lines.
-
...
INVITE sip:9500@192.168.154.135 SIP/2.0
...
The "192.168.154.135" is the domain name that the PBX searches. Obviously, it does not match the string "Testing1".
Just call the domain "localhost" if you want that all requests match that domain.
-
i looked at the call logs and i find that when i call from extension 2490 phone on the "domain1.pbx.com" account, it actually says the call is beign made from a diff domain.
then when i add my DID to that domain, the incoming calls work going to that domain
can it be that the domains are gettign mixed up once i deleted the "domain1.pbx.com" domain?
If you have just one domain, taking the "localhost" out is just causing these kind of problems. SIP is strict with the domain names, and if whatever is behind the @ does not match a domain name, the request is rejected. That is why we introduced the "localhost" magic word.
The DID matching deals with the problem that sometimes you really cannot tell where the call will go to and the DID are a hint which domain it should be.
Again, if you have only one domain, use the "localhost" name.
-
We have a number of fairly long dial-plans setup ad quite often we have a need to crreate dial-plan with the same entries as an existing plan but with some required changes for a particular user. At the moment we have to recreate the new plan from scratch and then make the changes. If we could copy an exsting plan and then make the changes it would save a huge amount of time!
What about having a way of exporting and importing them in a text format?
-
we re-installed the software on a new os on our machine
the problem ere having is that when we back up the directory and then put it on the new system, it doesn't OVERWRITE the existing accounts, it just APPENDS it, so local host stays there, and what was the 1st domain on the backup file, is being appended to the local host domain which adds all the default extension
does anyone know what else stays that I would need to delete? I had to delete all the default extensions
Yea, just rename the old directory and put the backup into a new, fresh directory. The only thing that you need to copy from the old one is the executable pbxctrl.exe, all other files can come from the backup. And well, copy the license key from the old setup to the new setup (copy & paste from the web interface).
-
Does anyone know or can post exact step by step instructions on how to change the email the system sends out on any option, like missed call, voice mail, new extension added, etc
You need the templates for that. It is not easy right now. Maybe we just add a web page where this content can be edited; I guess that would make most user's life easier.
-
I should say there appears to be one final issue with xlite/pbxnsip.
When xlite goes into idle detect, after that it appears that presence stops indicating right untill xlite reregistered.
Otherwise its fine.
matt
ps-let me add to my finding: it appears if xlite is in the autodetect and goes idle status and is there for several minutes it stops showing status and needs to be restarted (sip re-registered)
Is that something that can be fixed with shorter timeouts? There are two settings in Xlite for this in the presence tab...
-
You know me, put Service Flag status in the SNMP tables...
Not sure if that helps the secretary... -
Logging Flag changes would be good.
That is a good idea. We could just send an email to interested parties.
-
This is my design:
pstn phone line #xxx-xxxx ---> patton4114 pstn gateway ---> pbxnsip ---> patton m-ata ----> fax machine
-we want to use fax machine for incoming and outgoing faxes
-when fax machine not in use we use that line for calls
Do we setup in patton4114 pstn gatway to send the fax it detects to patton m-ata?
or do we detect fax in pbxnsip ivr?
If everything is in the LAN, I would keep it to G.711 and make life easier. Though using Patton on both ends might be easy.
On a side note, you can use the auto attendant to detect the fax tone and then send the call to the fax extension. A fax tone is like someone pressing "F" (instead of DTMF 0-9).
-
Sonus is sending this string in the invite and is supposed to be legal. Anything we can do about it since the pbx sends back a 404.
It is legal, but unusual. You can deal with it with the pattern !([0-9]*)!\1! or !([0-9]*)!\1!t! in the "send call to extension" setting of the trunk.
-
With pbxnsip help I've gotten xlite presence to work and that partially fills the need to indicate DND status...
But for those not ussing xlite it might be nice if PAC would indicate pbxnsip extension DND status...
Unfortunately there is no standard aboud DND. The IETF probably does noch spend time on such "trivial" issues... Maybe XMPP is the answer.
-
OK I figured it out. On the CS-410
If I added the port I wanted to use on the end of the SMTP Server setting it works.
Ex. smtp.comcast.net:587
However, I did try this in the version prior to 3.1.2.3120 (Linux) and I believe that it did not work.
As many of us know, when testing and observing, the results if unfavorable are sometimes forgotten.
Some servers run a encrypted service on ports like 587. That wount work. The PBX does support TLS, but first the traffic must be unencrypted on the regular service. Maybe that was the problem here.
-
This would all be great - we've never been able to make a scheduled flag flip from an extension. We have tested this on every version, included 3.2, since these many related threads were first started over a year ago. Screen captures or a wiki page would be great.
Hmm. We could introduce something that is a "manual override expire". For example, having something like "+1:00" would mean that a manual override would last only for one hour. Alternatively we could have a midnight event that just resets all manual overrides, so that at least in the morning everything is according to the plan.
-
Sorry to jump back to this 'old' message but i still do not understands why the PBX would wait until the next morning ?
I assume the Flag is set bij a 'status change message' with the desired state as optional parameter. I think this would the best way to program such a functionality. No matter how often or how much the status changes, it would do exactly what everybody would want it to do ?
What would be the problem in this setting ?
Well, AFAIK the latest state was: In automatic mode, the times define the transitions, not the state. In other words: At the end of the day, the service flags sets the flag - if if it not set already. That would cover the case where the secretary decides to home earlier. The same applies if she shows up early in the morning - if she shows up earlier she can anticipate the transition and turn the flag off before the service flag does that automatically.
The case where she stays longer is not convered. She would have to clear the service flag right after it switched on and then set it again manually.
-
We would like to use GTM+5 (Ekaterinburg, DST) timezone in pbxnsip, but there is no such setting. Advice please: it's possible to add manually support of this timezone?
We have Moscow:
<zone name="GMT+3"> <description>Russia, Moscow</description> <gmt_offset>10800</gmt_offset> <dst_offset>3600</dst_offset> <dst_start_day_of_week>1</dst_start_day_of_week> <dst_start_month>3</dst_start_month> <dst_start_time>02:00</dst_start_time> <dst_start_week_of_month>Last</dst_start_week_of_month> <dst_stop_day_of_week>1</dst_stop_day_of_week> <dst_stop_month>10</dst_stop_month> <dst_stop_time>02:00</dst_stop_time> <dst_stop_week_of_month>Last</dst_stop_week_of_month> </zone>
So the only thing we have to change would be the gmt_offset?
Not sure if that helps the secretary...
Failed Web Login Crash
in General Setup
Posted
If there is no need to upgrade don't do it... We keep the release notes on the Wiki. If there is nothing that interests you, "never change a running system".