Vodia PBX
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Any chance to get PSTN logging? Hint: You need to reboot the device to get them after enabling them in the web interface...
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... set the Log Watch List to 9 ...
Set it to something lower, e.g. 5. Then if you are logging on log level 5, 6, 7, 8 or 9 those messages will make it into the log.
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Unicorn 4102 is the IP Phone produced by Hanlong Technology Co,Ltd.
80×39mm LCD with blacklight，support 4 lines with 16 characters every line
2 line appearances with LED and 2 independent SIP accounts
dual switched audio –sensing 10/100Mbps network ports with integrated POE(option)
Support RJ-11 headset jack(option)
It looks to be provided with competitive prices.
Does it work with the pbxnsip PBX?
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Looks interesting. Is there any spec for this?
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Looks to me like you are talking to the wrong port (UDP 5061) on the Avaya system. Check your outbound proxy. Maybe you have to use TLS or change the port 5060.
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We have set up the integration with OCS using the Wiki page. Basically everything works fine.
I want to have different dial plans for different users for outgoing calls from OCS.
You can assign dial plans on per-extension basis. Is that the problem?
While setting up you have to set "Assume that call comes from user" to one of your extensions. This way all outgoing calls from OCS will have this user as caller-id and will use the same dial plan.How should the trunk be configured that it can identify the extension on its own?
Well, the problem for the PBX is that the call comes in on a trunk and it goes out on a trunk. Someone needs to pay the bill... (and that extension dial plan is being used). Not sure how this problem can be solved in another way.
The calls from OCS seem to come from +xxx@<fqdn-of-mediation-server>. I need to map xxx to a local extension.What about a tel:-alias? Does that solve the problem.
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Do you have the SIP packets (INVITE request and the response)? If you know the ISP IP address, just filter for that IP address then you will see the relevant packets and you can copy and paste them here.
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i want to give my general manager a direct line
meaning only he can access thatparticular line and also he shoulde be able to access any other line in the cs410 device
Make sure that you assign the real DID numbers in the PSTN gateway settings of the CS410. Then you can do inbound call routing as described in http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk. That part is easy...
Outgoing calls are a little more complicated. You can include the parameter "line" in the destination, then the cs410 FXO gateway will use only that line (for example, "line=3" means line 3). This way you can assign a dial plan to the executive that will send all calls to a specific line. Making sure that the other users do not use that line is a little bit more difficult. You would have to have three different dial plan entries and use the failover feature to find the first free line.
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i have a direct line connected to the pbxnsip
how can i do the follwing scenario
i just want a particular user to access this line
Line = trunk? So you want that only a specific user can use this trunk for outbound calls? Then you must create a dial plan specifically for that user and route the outbound calls over that trunk.
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the problems seems to be that if caller-id is not suppressed on the calling side we see an "auth requires" message in the sip logs on the inbound call coming in to the dialed number
Looks like the PBX believes that the call comes from an extension, not from the carrier.
One way to solve the problem is to take the domain context explicitly out and then route the calls just on DID basis. Which makes sense because usually incoming calls from the switch come from PSTN and then the PBX has to match it to the right domain based on the tel:alias.
That could be done by putting a different domain name into the outbound trunk that sends the call to the switch.
For example, you can set up just one global trunk to the switch that uses the outbound proxy and the domain name of the switch. That global trunk is in a dummy domain that has no accounts. Then incoming calls from the switch get distributed into the right domain based on the DID number (tel:alias). Outbound calls get the ANI (first tel:alias in 2.1) of the calling extension. When a loop from one domain to the other happens, the switch will send the call back to the PBX, the PBX will put that into the dummy domain, then look the DID up and send the call into the right domain.
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So is it a bad idea to use the Digium hardware with trixbox, or did I misunderstand you?
Using Digium hardware with Trixbox is not a bad idea. If you want to use the Digium card with trixbox, Asterisk will work as PSTN gateway, just like other PSTN gateways (e.g. AudioCodes, Cisco, Mediatrix, Patton, Parlay, Vegastream, Welltech). Other vendors of PCI-cards offer a SIP API (Sangoma, Dialogic and I heared about Junghanns), so that you can use the PCI card in the same server where the PBX is running on Windows. Digium's only API (correct me if I am wrong) is Zaptel, and the only PBX supporting that is Asterisk. pbxnsip does not support Zaptel.
So if I were to have a receptionist with an EXT module, could they have direct access to all the incoming POTS lines by mapping the lines to the keys?The problem remains the same, no matter how many keys you have. Grandstream uses the "BLF" (aka RFC 4235) standard, which does not permit a call pickup. In other words, the pbxnsip PBX can tell the phone that there is a call coming in, the phone might even be able to display a blinking LED, but when you press that key nothing happens. IMHO that is not usable.
We have specified a "buttons" package for Instant Messages that solves this problem. See http://wiki.pbxnsip.com/index.php/Buttons for more information. However, Grandstream does not support this method. It is always good if customers asking for features, so maybe you should ask them.
I would just use the "private" lines one the phone (where the phone assigned the LED to calls itself). That is much easier and most customers think that is okay.
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The GXP2000 says it has 4 line indicators, does that mean that if you have more than 4 incoming lines you would only have access to 4 of them? Is it possible to program the rest of the indicator buttons to connect to more than 4 external lines?
I will be using a digium TDM403E with Echo Cancellation Module, how difficult is it to configure trixbox to display caller ID from those POTS lines on the GXP2000 (or GXP 2010, 2020) displays?
In SIP, the term "line" has a different meaning that in good old PSTN world. In principle you can have as many lines as you want. Usually the SIP phone has its own was of mapping phone calls to LED. This has the advantage that the above problem does not occur; however it is not the same behavior people had with the key system. If the first LED lights up on one phone, that might be the second on another phone.
If you want to use Digium Cards you must use Asterisk to talk to the card (Trixbox is built on top of Asterisk). As far as I know there is no SIP API available for Digium Cards yet.
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We don't have PAC,as I understand it still in beta stage
Is it possible to check traffic between two extensions & TAP SIP messages using JAVA SIP APIs
Yes PAC is in beta, but the SIP messages between PBX and PAC are not. That is why you can see in "real life" how the PBX and the PAC interact and then do the same thing in Java.
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We are using 2 Snom 320phone in an agent group.
We we call the number the phones don't start ringing ad the same time.
There is an delay for about 2 seconds form the time that phone 1 starts ringing for phone 2.Is there somting we can do about this?
Well, that is probably a feature... The ACD tries to invite agents in steps. Inviting all agents at the same time put a heavy load on the system (SIP can be pretty heavy), therefore it is a good idea to slow it down a little bit. But you can change this, check out the settings in the agent group. Just set the number of agents that are added in each stage to 10.
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I need to build customized s/w with real time event monitoring on PBXnSIP
Can JAVA SIP APIs be used ?
Is there any way of doing this
Need help & guidance
That could be a good starting point. Maybe you can check the traffic between the PAC and the PBX. Essentially it is just Instant Messages. Then you can use the Java SIP to register and receive the IM and then build upon that.
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I've tested a little further, the issues is not really a beep, but something that sounds like a small drop in the sound when another call comes in.
Other SIP phones like the Siemens DECT/IP don't have this, so it seems a SNOM pbxnsip thing.
We don't want to start mixing the PBX audio in the case of a call waiting, this is (a) pretty complicated and (
the SIP model is that is the job of the SIP phone. I guess snom needs to fix this. -
operator console, i thought the extension was not licensed.
Currently the PAC registers just like any other SIP device. This is because the PAC behaves pretty much the same like the LED keys on a hard phone. That means you need an extension license (unless the system is "unlimited").
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I received the files to upload to the CS410 to enable it to be a DHCP Server (saves tons of $ in hardware for small installs since most small business grade routers will not send options 66 and 150) I edited/uploaded the files, set the server to run automatically, and changed the permissions all as per the readme file, however the CS410 is not responding as a DHCP server to requests on the LAN port, and I do not know what diagnostic commands to use to see what might be going wrong. Any suggestions would be appreciated.
Well, the DHCP server is just a "regular" Debian DHCP server... That means you can search for the respective documentation in the Internet.
One of the reasons why we don't include it is that DHCP servers require much more support than just a DHCP client...
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I tried the things you suggested except for the FXO signal booster (which I'm not sure exactly what kind of device you mean) and I still do not have Caller ID. I put off trying to fix it for a while as it isn't completely necessary. However, I have some time now and would like to see if we cam figure something out.
In my tests lately, I can only get the caller id to work after boot and not again. However, I noticed in the PSTN logging that there is a line about Caller id. After every call hangs up the line "PSTN: enable_callerid 0" comes up. Is this disabling the caller id after every call? If so, it would explain why it works the first time if the default of callerid is enabled.
No, that log message just says that it (re-)enabled caller-ID detection. Maybe you can try to incrase the log level? Please restart the box after that, so that the FXO subsystem can pick the change up.
Also, if possible use version 2933. That seems (so far) to be the best regarding caller-ID.
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hi its extremely difficult at this stage. moreover its in production right now, and handling calls so im afraid its impossible at this stage. so only im looking for a fix to this problem so that this problem gets solved.
I guess then the only choice we have is looking at the SIP packets and try to find out what the server does not like...
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1) microsoft speech server 2004 RC2
That is already 4 years old - in Microsoft country a lot happened since then. Any chance to upgrade?
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i face a production issue with MSS server 2004 R2. when calls come into the ivr during many cases the voice prompts are not played to customer and apparently xfers out to agents.
Speech Engine Services(SES) rejected the incoming audio connection for the following reason: SES was unable to accept the audio connection from <ipaddr> because the header could not be read. (Microsoft.SpeechServer.SpeechServerException). An existing connection was forcibly closed by the remote host (System.Net.Sockets.SocketException).
im unable to get any details from logs wat is causing the problems.
i appreciate any suggestions to resolve this issue.
Hmm. I know that a few folks got the Speech Server working, however I beliebe that the 2004 version might be a little bit outdate (but I am an the big Microsoft expert here).
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it's a mini call centre, at first when customer help the IVR, customer can select 1 to talk to operator, 2 to key in the Identity number. if customer key in the identity number, these will forward to a web database server for veirfying, i managed to set it up via SOAP, however, I am not sure how to setup to let the web server return a message which can be understand by pbxnsip ? apprciate your help please, thanks
Did you already check out http://wiki.pbxnsip.com/index.php/Linking_..._to_an_IVR_Node? Did it help?
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Is this a common prob?
No. The service must (and does) run without reboot. It is a PBX service, rebooting it every day is not an option.
If you need to restart the service because of one-way audio the service is still there, but cannot open media connection. This might be because there are no more ports available (check your RTP port range). The reason can be a lack of resources like ports, but it can also be a firewall that has (for example) too many TCP connections that need to be closed.
There is an delay for about 2 seconds form the time that phone 1 starts ringing for phone 2.
Configuring Polycom SoundPoint IP 430 w/pbxnsip
in Polycom Phones
Posted
So that file is not in the generated directory? Did you manually edit the pnp.xml file?
Check out http://forum.pbxnsip.com/index.php?showtopic=701. There you see the logic how the PBX searches for files.