Vodia PBX
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So we did build 3.1.1.3110 for all operating systems today and made them available from the pbxnsip software download page (http://pbxnsip.com/software). Although there are still a few open points, this should be the best version ever. Release notes can be found on the Wiki on http://wiki.pbxnsip.com/index.php/Release_Notes_3.1, and as you can see we did fix a lot of problems and also added some interesting new features.
The main point of 3.1.1 is that tel tel:-alias has been replaced with a telephone number. That makes it easier to represent that number in different environments, international or carrier-dependent and it also makes telephone number matches more safe. Also we added a new trunk identification mechanism possible that is very useful for multiple-domain environments. We made it possible to use the Cisco 7961 phone series with the pbxnsip in multiple domain environments, and even use features like park and pickup. And we found a ugly memory leak that could surface when Polycom phones are automatically provisioned with TCP transport layer.
Thanks for all those who were helping out testing the different builds in all kinds of environments!
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I'm trying to figure out a way to create a dial plan that allows 7 digit dialling within my local NPA, and all other LD requiring 11 digit dialling (i.e 1-NPA-NXX-XXXX).
Also keep in mind that the dial plan on the PBX has nothing to do with the question when the number is complete. You you can safely use the pattern "xxxxxxx" (7 digits) for all 7-digits, and then the replacement can just be 212* (if 212 is your area code).
Then later in the dial plan, if you don't insist on the length, you can use the pattern "1*" to match anything starting with 1, and replace it with the matched behind the 1 (you need no replacement for that).
So you dial plan could look like this:
Prio: 110; Pattern: 011*; Replacement: 011*
Prio: 120; Pattern: xxxxxxx; Replacement: 212*
Prio: 130; Pattern: xxxxxxxxxx; Replacement: *
Prio: 140; Pattern: 1*; Replacement: *
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[...]
INVITE sip:egix@66.158.175.178:5060;line=a87ff679;transport=udp SIP/2.0
[...]
[5] 2008/12/16 21:23:01: Trunk 3176604804 sends call to egix in domain localhost
[5] 2008/12/16 21:23:01: Trunk call: Could not identify user
In SIP the destination is in the first line (the Request-URI, see http://wiki.pbxnsip.com/index.php/Request-URI). That means the PBX is searching for the user "egix".
If you want to use the "To"-header, then you can use the following pattern in the setting "Send call to extension": !(.*)!\1!t!
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I am struggling with some caller-id issue
Sicne upgrading to 3.0, the caller id don't show correctly. I tryied to play with some of the options but can't make sense of it
HEre are some logs
Initial FROM HEader is Daniel/101 but subseuqent headers shows the real phone number?
Anybody has some insight?
The PBX is a B2BUA. That means that different Call-ID may have different From-header. Within a call, the From is not allowed to changed (unless the connected UA indicates it supports from-change, see http://www.ietf.org/rfc/rfc4916.txt).
What the PBX puts into the other leg of a call is a complex topic. "It depends". Trunks have their own rules, and the address book plays a major role for inbound calls.
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i have the following in my active calls!
2106/02/06 01:28:16 @ @
when i try to click the X to delete the call nothing happens...any ideas? im using version 3107!
For some reason the timestamp seems to be empty, as the from and to-fields. That is indeed strange. Any insight on what kind of call can cause this? Maybe click to dial?
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My MAC users are seeing this in the Subject line:
Subject: =?utf-8?B?W25leFNJUF0gVm9pY2VtYWlsIGZyb20gVW5hdmFpbGFibGUgKDgwMT Y0NzI2NzcpCg==?=
It works for my Windows guys just fine.. any idea?
There was a problem with RFC2047 headers (see http://tools.ietf.org/html/rfc2047). Will hopefully be fixed in the next version.
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You can create a custom dial plan that will add the 9.
That is just another reason not to use prefixes like 9. IMHO people should just enter the number that they want to dial.
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Been having issues with this new cs-425. I reset to factory defaults and it looked like it took care of the problem I was having (endpoints not able to register).
I checked it this morning and my domain is gone completely, along with the accounts. The static IP I gave the unit is still there, but when I go to create a new Domain I receive the message: "There are no more licenses available. The domain has not been created".
Currently running version 3.0.1.3023.
Ouch. Also on the CS410/CS425 you can make backup's of the file system. If you have a SSH client just log on the system (see http://wiki.pbxnsip.com/index.php/Installi...P-PBX_Appliance), in the end it is just a regular Debian computer.
Last resort is to send an email to support with the MAC address and ask for a new license key.
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The are using the polycom 650 and this issue only started on Friday when I updated the system to 3106...
Hmm. Can you upgrade the phone to 3.1.1 (see http://wiki.pbxnsip.com/index.php/Polycom on how to do that)? We have a 550 here and there everything seems to be working fine; however we are using 3.1.1.
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Can you turn SIP packet logging on? You should see a packet with the header "INVITE sip:xxx SIP/2.0" (see http://wiki.pbxnsip.com/index.php/Log_Setup on how to do that). Probably the SIP provider rejects the call.
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Hi this is a regular call...and it is not putting the call on hold...what is happening is a follows.
call number one comes in and the receptionist picks up and is talking to the caller. then a second call comes in and instead of just hearing beeps all of a sudden she cannot hear the original caller anymore but the caller can hear her! the only way she can get audio back for call number one is to put the call on hold and then resume the call!
What phone are you using? What firmware version? At first glance sounds like a royal bug on the phone.
Is there NAT involved? If that is so is there a "smart" router that screws the calls up?
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i have an offic with approx 90 users...what is the best way to setup the paging and how do i go about it?
i currently have them set on UNICAST but recently when i hit the extension number it just gives a fast beeping sound...i look in the logfile and it says it cannot find the extension but then says that paging would cause too many active calls so im guessing i have to go to multicast.
is there any easy way to set this up? my phones and server are using an ip address of 10.0.10.XX
Unicast is not a good idea with 90 users. You can use the multicast mode which sends the RTP to a IP address/port without sending out INVITE packets. If that IP address is a multicast address (starting with 224.x.x.x), interested devices can play it back. snom phones do that and if you use Plug and Play then they are automatically set up for the first multicast paging group in their domain.
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I have connected the Broadband modem to the WAN port of CS410 with the following setup
Broadband modem with PPOA with the Fixed Public IP to the wan port of CS410 and the IP phones / PC connected to the CS410 LAN port through switch.
From the PC I cannot browse the internet although I have put DNS IP in the CS410.
The CS410 is not a router... It is just a computer with a couple of IP addresses that runs an application called "PBX"!
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If i ad an extension my phone keeps connecting to the wrong domain.
After i removed the deafult enabled domains my phone does not connect anymore
Not even if i add my domain name like 40@......
The name "localhost" is like a wildcard. It matches any name. That makes it easier to use the PBX in a environment where there is only one domain. Once you have more than one domain, you must use exactly the name of the domain in your user agent.
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no this is different... now i have muliple domains sending TCP calls to MS Exchange.. and it is only sending the NIC IP.
Didnt you say TCP IP's were working?
Well the TCP is for inbound TCP traffic. When the PBX opens an outbound TCP connection (PBX being a TCP client) then the OS will assign the IP address; and the PBX has no control over it. AFAIK you cannot bind a TCP client socket to a specific address.
IMHO client authentication based on IP address is not the right way to solve this problem...
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MY RECEPTIONIST IS COMPLAINING THAT WHEN SHE IS ON THE PHONE AND A SECOND CALL COMES IN, IT MUTES THE CALL SHE IS CURRENTLY ON SO SHE CANNOT HEAR ANYTHING UNLESS SHE PUTS THE CALL ON HOLD AND THEN HITS RESUME!
Is that a regular call coming in? Or is it a intercom call? The PBX cannot put an ongoing call on a phone on hold, only the phone can do that. There was an issue with a Linksys phone some time ago; Linksys argued this was a feature, because the intercom might be an life-or-death call. Not sure if they changed their opinion in the meantime.
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Inside of my extension settings I set my mobile phone number. When I make test call from another one extension to my - both phones (ip-phone and mobile) are ringing at the same time, as configured.
But when I trying to initiate Click to Dial with my username and extension - the system calling back my IP-phone _only_, and did'nt tryed to reach mobile phone number.
Advice please, it's possible in general?
Yea, at the moment that is the intended behavior. If you want to get a callback on your cell phone, consider using the callback feature (see http://wiki.pbxnsip.com/index.php/Calling_Card).
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I have setup PBXNSIP but i can not find the SMTP server settings, not on the general settings leven and/or domain settings.
What am i doing wrong ?
On system level they are hiding under the logging tab. There you can also turn and and off if the domains all use the same settings.
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I think we have a interesting new device that the PBX can talk to: The FritzBox now support SIP trunking, see http://wiki.pbxnsip.com/index.php/AVM. This is a device that can talk to ISDN, primary in Germany.
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so when i have login/logout button the LED does not change automatically. ie until there is an event.
So I press login the light does not turn on until a call comes through the queue.
I tested that with version 3.1.1.3107 and there is seems to be fixed.
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The counter for the OID 1.3.6.1.4.1.25060.1.1 (calls) seems to be broken. After reboot it is fine but over time it presents a growwing off set in number of calls.
Well, that is not a good sign. These calls are probably really in the internal database. This can happen when there is some SIP problem (handshake problem), but those calls should clear up after some time (8 hours or shorter).
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Debian revision? Also, please address the logging issue? Lastly, I still have not had an answer on forcing outgoing ANI changes as currently it sends the incoming ANI out for redirected calls and that is always not possible as some vendors make your ANI be one of the known DIDs?
The latest & greatest can be found at http://pbxnsip.com/protect/pbxctrl-debian4.0-3.1.1.3107. If you want to force a ANI outbound, set the trunk Remote Party/Privacy Indication to "No Indication".
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I was wondering if there is a way you can send call accounting CDR data on a per domain basis. The senario I am thinking of is setting up a hosting company, and a couple of the customers want to run their own call accounting software. Right now if I point it at their server's IP address, it works fine, however they can see all of the data for all of the domains. The second problem is that if I have multiple clients, I will need to be able to send to multiple different IP addresses of call accounting servers based on the domain.
The big question is how they want the CDR. For simple CDR I could imagine introducing a domain-based setting that overrides the global setting.
We'll soon have CSV CDR written to the file system; the location may include the domain name. That might be possibility to solve the problem. If you are running a SQL server then we might also be able to feed the CDR into a table. We could also introduce a domain-based setting here.
And of course, you can send emails with CDR per day. That is there already for some time.
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One of the things I like about pbxnsip is how crisp and quick the pace is for menu options when you access voicemail ("This is the main menu of the mailbox.<pause>To hear your messages, press 1.<pause>To ...")
These pauses are nice and short. But when I call the auto-attendant, there's a much longer pause between options ("Press 1 for sales.<pause>Press 2 for accounting.<pause>Press 3 for support.").
It would be awesome to be able to control the pauses between options especially on the auto-attendant since I think voicemail is perfect.
Is there anything I can do to override the default behavior?
No, at the moment this is hardcoded to two seconds. Maybe this is too long. What would be better? One second?
3.1.1
in New Features/Versions
Posted
Ops, seems the link was wrong... Should be okay now.