Vodia PBX
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Version is 2.1.12.2489 (Win32), Under Extensions Registrations; Field: Bind to MAC Address
Okay.
Incoming calls have nothing to do with MAC addresses. The MAC address is used only for plug and play. Then once the phone is configured it will probably keep the registration alive and that's why the PBX sends the call to that phone.
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Has anyone had the experience that if a * was entered and later removed, that the system will still think it is there and connect incoming calls only to that extention: even after removal. I am having general problems sending the calls to the endpoints where they belong.
Where did you enter it? Into the MAC address field? What version?
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I have been rummaging around and have not found a step by step guide to adding extentions. I am having problems with MAC and ip addresses not registering. Has anyone had a simular experience?
What did you do? Did you create an extension in the domain? How far did you get? Maybe http://wiki.pbxnsip.com/index.php/Creating_New_Accounts is a good starting point.
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any one who can help me how to set-up lines (NOT CO-LINES) , but matching LINE setting from the Snom to the pbxnsip button.
I know how to do it directly on the SNOM phone, but who do I make it work with auto-provisionering of the buttons.
Well, definitevely check out http://wiki.pbxnsip.com/index.php/Snom and http://wiki.pbxnsip.com/index.php/Assigning_Buttons.
You can select "shared line", but IMHO that is legacy from old key system times and you never get customers happy because their old 1976 system works in a silightly different way... Better "upgrade" them to a PBX functionality and use private lines (managed by the phones) and park orbits instead. Then you can also deal better with a mix of phones that have only two lines and others that have much more.
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I'm having a problem with getting reliable inbound connections.
Hardware/software details:
PBXnSIP v2.1.14.2498 (Win32) (Win XP Pro)
Sonicwall TZ170 - firmware SonicOS Enhanced 3.2.3.0-6e
Linksys SPA942 Phones - firmware version 5.2.2(a)
Linksys WRV 200 Routers - Firmware 1.0.39
VoIP provider is Callcentric
VoIP Layout Details:
The Sonicwall is the main router with the PBXnSIP hanging off of it. The Sonicwall is configured to pass all PBXnSIP ports to the PBX PC. There are 2 branch offices that connect via a VPN provided by the Linksys router to the sonicwall. Each office has a Linksys phone.
VoIP problem:
When no inbound calls are placed for an undetermined about of time, the first inbound call is either unanswered after a long silence or is answered by CallCentric's voicemail. If a second inbound call is placed right away the PBXnSIP will pick up the call in a normal time. All incoming calls for the next undetermined amount time, are answered as normal.
Trouble Shooting details:
This problem appears to be corrected by enabling SIP Transformations under the VoIP section of the Sonicwall. However, when I enable this, both branch offices will lose audio both ways. The phones are still communicating with the PBX because missed call notifications are e-mailed when attempting to call one of the branch offices.
Any help you can provide will be greatly appreciated.
http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems mentions a few search tips. Maybe it also makes sense to track the registration of extensions (send out an email when the status changes) to see if the registrations are stable.
SoHo-Routers can be a problem. Maybe if you have a problem next time try to reboot them to see if that makes any difference. We even had cases where the Ethernet switch gave is a problem and needed to be rebooted. "Divide and conquer" can help to isolate the problem.
Of course, if you are able to see where the INVITE from callcentric goes then you should be able to pinpoint the problem. Maybe it makes sense to run Wireshark at critical points. If you filter for port 5060 then the file size should be not outrageous.
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Thanks for all your help. I have tried everything to get this to work and for some reason couldn't get it going. I was looking through the forums and came across a solution in another post. The user mentioned that he had the same problem and ended up changing the pbxnsip domain back to localhost from the fqdn domain name he was using. Since I had tried everything else, I gave this a shot too and voila it seemed to work.
Oh yea, that is a good point... But then make the "localhost" a alias name, so that if you get a call for something else than the primary domain the PBX would still take it.
The reason I was looking through the other forum posts is because during my testing a found another error with my setup. When I dial into the pbx from an external number and pickup the call on one of the phones, I can hear the audio from the remote phone but they can't hear anything from me. If I dial out from the PBX this works fine. I know that normally this is considered a NAT problem but this used to work fine and nothing with my NAT setup has changed. I have an access-list on my firewall allowing everything through to the PBX and because I have several public IP addresses I have forwarded an entire IP through to the box so there should be nothing blocking it at all. The only thing that has changed is that I have upgraded my CS410 to the newest firmware software release.The logs see the calls being established but don't give any indication as to why they don't go through.
The newer versions check your IP configuration in a JavaScript. There are a couple of combinations that screw up the default gateway in Linux. The script checks those when you hit the save button.
In the newer software releases there is (AFAIK) no change regarding the handling of NAT and IP routing.
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Hi, I have a Siemens C470IP. (I'm using PBXnSIP 2.1.14)
When enter the mailbox it asks for the pin.
I entered the correct one but it says wrong code entered.
I saw some settings in the phone about DTMF. (I read the wiki but I have a different screen, I don´t have the DTMF option like in the wiki)
DTMF over VoIP connections
Send settings: Audio, RFC 2833 or SIP Info (I can select or deselect none, one or more options)
When using G.722-Codecs (wide-band connection) DTMF Signals cannot be transmitted over audio.
Software of the C470IP :
Firmware version: 021230000000 / 043.00
EEPROM version: 121
What do I need to do to get this one working?
The PBX does not do anything special for G.722, so I would say that is not the point here. Do you have any chance to get a Wiresharl trace? Then we can see if there is a problem on the media level.
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I see the option to set who the call comes from. You're right, I didn't have anything set in this option. I set this option to one of the PBX extensions but I am still getting the error. I do have redirect enabled on the trunk. Does it matter which extension I use for this billing option and do I have to restart the pbx after I make this change? It seems like a pretty simple option but I am still getting the error that I don't have permission to transfer to the extensions.
Well, make sure that this extension can really dial that number - check the dial plan what was assigned to the domain and the extension.
No restart neccessary for this.
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Well, currently the PBX does not support TLS for email... Short term workaround is to use another SMTP email relay that uses plain SMTP on TCP.
Maybe this is the opportunity to start doing this!
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Ok, I am fine with hitting the checkmark button on the phone when I am making calls for now. All my phones are Snom phones so I will see what I can figure out about this on their site.
The only other issue I am having with this Exchange setup is related to the Exchange Auto Attendant. I want to use Exchange for the AA because it supports dial by name and I have the mailboxes there. I setup the VoIP line to automatically go to the auto attendant of the Exchange server. When I dial into the line the Auto Attendant answers fine. The problem is, when I try to enter any extension to connect to a phone I get the error "We are sorry but you are not allowed to place this call". That happens regardless of which extension I dial through the system. The Exchange AA also gives you the option to press # after you dial an extension to go directly to the users voicemail. If I do this I transfer successfully to the mailbox so it seems like the routing within Exchange is working but I am not routing back out to the PBX.
I have gone through the forum and the wiki looking for a solution to this but haven't been able to find any configuration information for working with the Exchange Auto Attendant. I used to use the PBXNSIP AA and it worked fine but would like to use the Exchange one now.
Oh did you put into the trunk an account that "pays" for the call? This is called "Assume that call comes from user" - and that user must have a dial plan that allows dialling this number. And of course the Exchange trunk on the PBX must "accept redirect".
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I do have one question that isn't totally related to this but does apply somewhat. I now have the PBX setup to require the user to press the checkmark to dial any extension and that is ok for the most part but I have tried to get dialplans to work so that they dial the moment the number is typed. For example; if I dial 7100 I would like it to dial the moment the last digit is entered. The same thing goes if I enter a users extension, I have to press the checkmark to make the call go through. It seems that by changing the default scheme setting the phone never automatically dials and I have to press the checkmark to make the call go through.
I have gone through the dialplan documentation but can't still seem to figure this out.
Technically that is a problem of the phone now. The "dial plan" of the PBX has nothing to do with that. So the first source is to check the phone vendor documentation.
For a few phones, the PBX automatically generates a phone dial plan (e.g. snom). The generation of this plan depends on the domains "Default PnP Dialplan Scheme". "User must press enter" means that there is no dialplan for the phone, the North America (NANPA) styles generate a dial plan for internal calls and eleven-digit calls.
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I have been looking at it now and the issue I can see so far is that the dialplan doesn't appear to be connecting to the Exchange server. I have a very simple dialplan that says 7* -> * out the Exchange trunk. This has always worked in the past. The issue I am getting is very strange, when I dial 7100 to reach my extension, the phone automatically dials when I get to 710 and doesn't wait for the last digit. The logs do say that the call is being sent out the Exchange trunk but obviously the extension doesn't exist on the Exchange server. I have gone through the entire setup of the PBX and can't find a problem. I even reset the PBX to default and reset both the Snom phones but after setting it up again I am still getting this problem. This seems to happen whenever I dial a number that starts with 4, 5, 6, or 7, they all dial after the third digit. I use 9 to get an outgoing line and it doesn't dial until I press the checkmark on the Snom phone.
You probably selected a dial plan scheme for the domain and then provisioned the phone through the PBX. Select "user must press enter" as scheme and reboot the phone.
The other thing is that you don't have to dial 7100, you probably should just try 8100. Then the PBX will realize the call should go to the mailbox, and then it says "hey there is a redirection to 7xxx" for the mailbox. The point here is that the PBX includes a redirection header in SIP that Exchange needs to work properly.
Other than the 9* dialplan to go out through the VoIP trunk and the 7* dialplan to go out through Exchange there are no other dialplan entries. Both the CS410 and the Snom phones are fully updated to the newest versions of the software. Should I roll back to an earlier firmware on the CS410 or does anyone have any other suggestions why this might not be working?No don't rollback, the latest and greatest has the best support for Exchange.
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It would not work. We have 5000 short dials we need to use. 2 digits will not help. Even something like using a # for shortdials rather than the * it might also work.
5000? That sounds like a really huge family to me. Anyway, then lets say we accept up to four digits.
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That really sounds like a lucky number! And a few important bugs have been fixed. Check out the downloads at http://www.pbxnsip.com/software.
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Using CentOS 5.1 and 5.2 on different servers. It is the same as RHEL5.
Give http://www.pbxnsip.com/download/pbxctrl-centos5-3.0.0.2991 a shot!
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Tried that ..... Error message is below. The error occurrs when I try to load the MIB into the management software.
Error parsing MIB:PBX-SNMP_mib
Exception while loading MIB:Could not parse the file PBX-SNMP_mib The error occured at the line no: 18 ,column: 1. For the table and row OBJECT-TYPE construct the ACCESS value should be 'not-accessible'. For other constructs the ACCESS should contain one of these values : "read-create , not-implemented , accessible-for-notify , read-write , write-only , not-accessible , read-only "
Eh? It is "read" right now. Not sure if "read-only" is the right way to say it.
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what will be the code for Codec G723? in wiki only tell codecs ulaw (0), alaw(8), G.722 (9), G.726 (2) or GSM 6.10 FullRate (3).
G.723 is not supported by pbxnsip. This codec is not worth it in a PBX environment:
- Bad audio quality (well, what can you expect from 5.3 kbit/s?)
- High CPU load (big problem for central network equipment that does support MoH and call barge in, recording)
- Show-stopper license terms (lots of patents, unclear situation who will take us to court us if we support it)
And don't forget that 5.3 kbit/s does not mean that a RTP session takes 5.3 kbit/s. The RTP packet overhead is somewhere in the 12-24 kbit/s area, depending on the packet size. Yepp, that's 3 times higher than the compressed audio information itself. Making G.723 even more pointless.
- Bad audio quality (well, what can you expect from 5.3 kbit/s?)
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I would try turning polarity detection off and busy tone detection off. There must be something that makes the FXO subsystem believe the call is disconnected.
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So have you had any PBXnSIP performance issues on a server with multiple domains and recordings being emailed to the customer? These recordings would be fairly large in size with recording times going to 60 minutes. This has become a major sticking point for one of our customers.
Well in the previous version the thread that did the SIP processing also assembled the email including the WAV. Because that WAV has to be base64-encoded, this could take some time. That is why we did not send long WAV files. Now the picture changed. Because there is the email thread which can work on this for minutes there is practically no more limit for the length for WAV files and sending out the recording of a two-hour conference becomes an option.
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1. I attached the section of log at the time of the drop - does this information indicate a *9 command?
Unfortunately, no. But what we can see is that the PBX does disconnect the call (sending BYE, not receiving).
We added a log message for the *9 disconnect. Maybe that is easier than trying to figure out what the problem was. What OS are you on and what version? Maybe you can try a build that includes the log message and we figure it out.
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We added the following line to the init.d script to create the pid:
lsof -t -c pbxctrl > /var/run/pbxctrl.pid
This will suffice until release 3.0
What OS are you on? Maybe want to try it out?
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I installed IPTABLES and set it up to allow only ports 22(ssh) and 5060/5061 on both tcp/udp, 8085/8086 on tcp and the entire RTP range in udp.
Do you see any problems with this and pbxnsip?
Has anyone else used IPTABLES and pbxnsip V2?
I V2 I would not recommend to play with IP tables. The PBX uses /proc/route to figure out what it's IP address is when sending requests out and iptables can very easily screw that up. In V3 it is a little bit more relaxed, because the PBX asks the OS for the routing tables through a system call.
And as in general with iptables, you really really really have to know what you are doing here. This stuff is not for beginners.
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Has there been any progress in this area? Is the WIKI up to date? We would like to have the ability to monitor stats through our NMS.
Well we proposed a MIB, but nobody responded - http://forum.pbxnsip.com/index.php?showtopic=112.
DTMF - SIP info
in Trunk Setup
Posted
The support is only for receiving DTMF INFO, no transcoding between signalling layer and media layer. This is just because we don't like to make our live miserable. Think about someone sending RFC2833 (=RFC4833) DTMF tones and then while the tone is playing back someone sends a INFO as well.
Plus the support for RFC2833 tones is pretty good these days.