Vodia PBX
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i want to establish pbxnsip with turkish prompts. but the turkish prompt file can not be downloaded. i think it is missing.
No need to record them. I think the link was wrong, try this one: http://pbxnsip.com/download/audio_tr_30.zip
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I replaced the pbxctrl.exe with the new one and I selected required options on the ports and trunk configuration page.
Unfortunately, PBXNSIP is still trying to change the codecs during the early media.
Try locking the codec both on the trunk and in the Ports section. Maybe the restriction in the trunk is not enough (maxbe that option is kind of superfluous).
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We have numerous networks and routes inside the routing table of the OS and also a few ips so the PBX is far from an optimal location to specify routes .. can it just be enabled on sip replacement list as well since you do that based on that info already or at least at an option (checkbox) to enable that function for the sip replacement list for use of the replacement IP in generated configs?
Okay, will be in the next version. What OS are you using? Would you try it out?
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will this new version better control DND? most of my problems is someone pressing DND button twice fast and it only sends 1 update to pbxnsip..
so the phone ends up not being on DND and pbxnsip has the DND flag on.
Yes that is a well-known problem. IMHO it is fine if the DND key is provisioned as button. The button profile must contain the dnd key, that might be the problem in real-life deployments.
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In the attached zip file there is a sample of a successful transmission from Patton to Zoiper, and a sample of a failure through the pbx, same configuration of Zoiper and Patton.
Patton = 192.168.101.7
Zoiper's PC 192.168.101.213
PBX 192.168.100.3
I set the Patton to repeat CNG three times
Really strange. Attached is the RTP in the unsuccessful case. If Zoiper is supposed to detect the CNG? I can hear it loud and clear.
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For a small percentage of our outbound calls each day (about 3 or 4 times per day), we'll hear ringing and then suddenly a fast-busy sound and the call drops.
Our SIP trunk provider is Broadvox and they did a capture of the problem when it occurred. What they said is that sometimes pbxnsip 3.2.0.3144 (win32) isn't sending a response to their 200 code to indicate acknowledgement of the connection. As a result, Broadvox thinks the caller on the pbxnsip side hung up so they disconnect.
Is anyone else running into this? If so, what version are you running (and what OS)?
You mean the ACK does not make it? What transport layer? I guess UDP? Maybe it makes sense to run Wireshark on port 5060 and record the signalling. It could be a problem with DNS SRV or the branch tags.
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Hello, I'm testing T38 for our costumer, a lot of them need to route a fax call through an AA.
I installed a Patton Gateway, configured to transform analog faxes in T38, then I have a Zoiper soft client on a machine, this client is an extension of our production pbx.
I switch the PAtton between two configurations, the first sends the fax directly to the IP of the Zoiper client, the seconds sends it to the pbx, this is the only difference between the two (infact, I only change manually the SIP proxy address).
pbxnsip has the inband detection enabled and an AA with the "F" detection, and routing works well.
now:
If I send faxes directly to zoiper it receives them and wireshark shows a T38 communication.
If i send the same fax through pbxnsip zoiper tells me the phone is ringing, as if he can't understand it's a fax, or a T38 communication.
Unfortunately the first case is not a solution, because i'm sending ALL connections to Zoiper, so in a real environment my pbx don't receive any call.
I think that pbxnsip in some way "strips" the ced tone, or sends it where it can't be heard by zoiper (I tried sip-info and rtp mixing them on Patton and Zoiper both), please is there someone knowing which are the correct parameters?
The CNG tone is repeated a couple of times, the receiver should be able to sense it... There must be something else.
Can you get a PCAP trace and put it somewhere where we can pull it down and take a look?
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I just tried, *90
it asks "please enter the extension number"
i dial "ext"
it says, you are not allowed to place this call.
If you like, please try http://pbxnsip.com/protect/pbxctrl-3.3.0.3159.exe. Please use the PnP mechanism to make sure that the phones get the right setup. Also make sure that the PnP firmware version is the 7.3.14.
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TIP: Reserve UDP ports from windows' own services to prevent signaling & audio
problems with pbxnsip.
UDP is a big problem nowadays under windows. With the latest DNS patches to
Windows machines ports used for UDP SIP and RTP audio can be eaten by
Windows' own DNS client and server processes.
Listed below is a registry key that will prevent windows from using UDP ports in the
same range that pbxnsip requires for operation.
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Tcpip\Parameters\
ReservedPorts=
5060-5061
10000-12000
Separate values for ReservedPorts using enter to goto next line. You may change
reserved UDP port numbers to whatever ports above 1024 are required to operate
pbxnsip in your environment. In the sample above the first line is for inbound sip
on 5060 thru 5061 and RTP media using ports 10000 thru 12000.
Vulnerabilities in DNS Could Allow Spoofing (953230)
http://www.microsoft.com/technet/security/...n/ms08-037.mspx
You experience issues with UDP-dependent network services after you install DNS Server service security update 953230 (MS08-037)
Whow that is very interesting stuff!
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All good, got it working. Removed "Send to Extension" from trunk and added aliases for DID numbers in Hunt group and all seems to work. Thanks.
Great!
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Today we had a client PBX with park orbits and low or easy SIP passwords get remote registrations from a Canadian IP address and the clients PBX was making outbound BANK CARD scam calls.... Caught it early but the lessons are clear and a few best practices are coming from the experience and perhaps a feature request...
Lesson 1. Complex Passwords are a must - No longer can we make it easy for the users
Lesson 2. Park Orbits will not enherit a default dial plan
Lesson 3. enable more logging an email notifications on extensions
Possible Feature requests - (optional allowable IP Address ranges on an ext Basis for phones to register from.
Cheers, and learn from the experienced.
Totally agree.
We introduced the script that checks password for their "randomness". Unfortunately, due to a request from the sales front, we were asked to disable it in the default installation, so that a password like "secret" is accepted as a password (like "", the empty string).
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"DND is still available through the Menu option"
With this i meant that DND is still available through the personal web-interface from PBXNSIP, so i assume it is not possible to block it through the phone.
I think it should be managed by PBXNSIP, is this possible ?
Well, the DND option might be still available in the phone. However if it is read only, then the user cannot change it.
The next version should contain a PnP that actually redirects the settings key to a XML screen, so that the user does not get to the screen at all any more. There is also has a DND item, and that could be controlled by the PBX.
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How can the MP118 FX0 be configured so that each PSTN port is addressable as a separate trunk? I want to configure Snom 360 phones with SLA and want to associate each button with a specific line.
Hmm. Difficult topic. Ideally the PBX would register the trunk on the gateway like it would do with a service provider; not sure if that's possible. In gateway mode, it is very difficult to find out from what trunk a call comes in (they are all using the same IP address). And it is difficult to tell the gateway which line to use on an outbound trunk. Maybe the Audiocodes has a special parameter for this.
Does the AudioCodes manual say anything about how this should be done?
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The cisco PNP give this:
domain: 65.181.50.254
and the cisco phone doesnt seem to like this. it keeps trying to register to the proxy1_address.
Yea. The Cisco phones seems to be a little "picky" with the provisioning. I believe jumbly knows the trick how to set everything up correctly for a multiple domain setup with Cisco phones.
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Hi, just new to setting up pbxnsip on cs410 and all seems ok so far except we have 3 DID FXO lines that I can't get to be recognised as different inbound calls in the trunks.
We want each FXO line to a different Hunt group extension so we can set different ring tones for each line etc.
No matter what line we call in with it seems to pick the wrong trunk and therefore the wrong hunt extension.
I have read the wiki on assigning DID numbers for FXO but it doesn't seem to match the version we are using which is V3.
Thanks. I hope I have described our issue correctly.
Well, essentially you can set the DID numbers that the FXO should report for the respective line. Then on the PBX you can use the inbound trunk rules to locate the hunt group. Probably it should be sufficient to assign the right alias name to the hunt group. Another obstable could be to choose the right 10- or 11-digit format.
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Is it possible to Block the use of 'Do not Disturb' in PBXNSIP ?
We have a few users who like to use DND while company policy is not to use it......
I managed to disable the DND button on the SNOM phones but DND is still available through the Menu option.
You could provision the DND setting on the phones and set it to "read only".
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Those ports are directed to the PBX by the modem. They are not directed elsewhere. I have no problemwith the SIPgate connection. It is registered. My problem is with the ISDN lines not anything else. I do not know if the city code must be included as well. I still think it is a problem with the Trunk. If I do not make progress I can reinstall the Modem from scratch. Ugh!
Did you set the country code for the domain? If that is the case, the PBX tries to be smart about the numbers. If you put a "49" there and use the area code (e.g. "40"), then a number like 00494012345 will be interpreted as "12345". Try to use the DID numbers in the way you would dial them from the phone. Maybe you have a problem with the matching of the numbers. In this case, you should see the INVITE packet coming to the PBX with the number (do you? what do you see?).
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Can you tell me when to expect this version 3.3 ?
All my phones are already on the 7.3.14 firmware so then I know if it would be better to wait for version 3.3 or to downgrade all my phones to firmware 7.1.39.
Release notes are not out yet. I think the biggest changes are Intercom and warning about the default password. Maybe also the way VLAN is being provisioned.
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NO same Router. Did you look at the info I sent regarding the trunk? It could have something to do with the way the trunk is set up. Perhaps the way the code is being presented. Or is it stopping at the router?
Do you have a chance to see where the router sends packets that come to port 5060? Maybe there is another device in the network that also uses port 5060 and opens a connection to the Internet - and then that port 5060 is taken already by the other device, not the PBX. Many routers have a way of seeing how the ports are allocated.
If you are registering a trunk to a service provider, you will also see the real IP address and port in the Via header of the response.
[Did I mention I cannot wait for IPv6? No more of these NAT problems.]
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Again when I dial in using my mobile I get no tone. On the pbx there is no record of a call.
How should the router be set up differently?
From what I read in this topic, you had to replace your old router with a new one and since then íncoming calls don't work and more. That tells me that something in the setup of the router must have changed. Maybe it is something simple like the PBX has a new IP address and the DMZ settings must be adjusted accordingly. But it might also be a problem like the new router is suddenly SIP-aware and creating a compatibility nightmare. It is hard to say from here what the problem is. If I had access to the router, I would check the DMZ, the provided IP address (DHCP) and if the firmware is the same, and if there is something else suspicious. I would also check the log file of the router for messages.
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I am looking for a simple RTP server that I can run locally and have PBX listenning to it something like iTune would be perfect!
Maybe also take a look at http://www.videolan.org/vlc/streaming.html.
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Under the main settings>general tab, in the Appearance section, there is a Compress Recordings option with a yes or no variable. Is setting this to yes compressing like the previous versions did or is this some sort of further compression going on?
No the compression is still using the GSM codec. IMHO a reasonable combination of CPU load, audio quality, Windows compability, and file size. If you don't compress, the codec will save 16 bit/sample.
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What should I do now in what order?
Check if the router sends SIP traffic to the PBX upon an incoming call. If that is the case, check the log of the PBX why it would reject that call. If there is no SIP traffic going to the PBX, check the router setup (again).
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Is there anybody who can tell me what Snom 300 and Snom 320 firmware would be the best when using with PBXNSIP 3.2.0.3143 ?
Should I use Snom firmware 7.3.14 or is better to use firmware 7.1.39 and can you tell me why ?
In version 3.2, it is 7.1.39. In version 3.3, we'll switch to version 7.3.14. The 7.3.14 version requires a different way of provisioning, some details have changed (e.g. the enabling of the intercom feature).
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I assume you put the source IP address in the list of SOAP trusted IP addresses? Do you see this in the LOG "[5] SOAP: Need a professional license" or "[2] SOAP: Request from untrusted IP address"?