[Issue with outbound from (caller-id)]

My local vendor for our PRI does not allow outgoing caller-id to be anything but one of our DID's .. the rub is we want that to be the case when calling out from an extenstion so we have opted for Remote Party/Privacy Indication: of an RFC one or the other .. which one someone calls in but I want to forward to a cell phone number, etc the PBX automatically sends that as the caller-id. What can be done about this to retain the proper outgoing DID for the ext the call was sent from, etc?

 

Well, the rules are defined in http://wiki.pbxnsip.com/index.php/Outbound_Calls_on_Trunk.

 

In the next version we'll have a new setting called "ANI" for every account, where you can explicity put the caller-ID that you want to this specific account.

 

I think we need to write something up that explains how this whole Caller-ID presentation works in theory and in real ITSP-operator life.

[Problem Receiving Calls on Aastra 480ict]

I have a Aastra 57i and having trouble getting it to provision properly, it will never register? Is the profile in 2.1.10 correct for this phone? Also, long ago you noted putting up all the config references for PNP in the html directory, any idea when those base configs or at least a reference to them may be available?

 

Did you turn password provisioning on (admin/ports/tftp)? By default it is off - because tftp has no security.

 

I am not very happy with the way the provisioning files are "moderated" at the moment by us (pbxnsip). Maybe we should have a public CVS server or a Wiki where everybody can edit those files, so that we can share the best configuration.

[Record Location seems to be ignored when call is manually recorded]

If I dont enable any automatic domain level recording options,

but want to manually enable recording via the Record On Key, Record Off Keys.

Then the Record Location syntax is ignored and the recordings are only stored in the recordings directory as msg**.wav

 

I would like to see the same Record Location syntax used - for these manual recordings as well.

(Or perhaps a second Record Location option for these)

 

Well, at the moment that is a feature. The user-initiated recording is supposed to be for recording only short parts of a conversation, like someone saying a telephone number of address or something that you don't want or can't to scribble down while talking. For these purposes it seems reasonable to put that into the (private) mailbox of the user, and possibly forward it by email.

 

What is the use case for recording to the directory?

[Address book]

Hmmm, only the first 9 in this case! Maybe I need to latest and greatest version, I think they're a couple of revisions behind..

 

That is a feature! Imagine you have 1000000000 matches.... The first 9 just fill the display, then you need to enter more digits (1-9) to refine the search.

[BLF & Hunt Groups]

We have 6 Grandstream GXP2010 phones (with an extension each) connected to PBXNSIP v2.1.10.2472 . Everything is working great, except for some strange issues with BLF lights when a Hunt Group is ringing. Basically we have set up the 6 phones so that the BLF indicators on each phone show the status of the 6 extensions - this works successfully for a direct call to an extension, but if all extensions are ringing (they are all in a hunt group) then the BLF lights go crazy - sometimes they all flash (which is correct) sometimes one line flashes, sometime two etc, and it is different on all phones! Even worse is that if the call connects or terminates, then some of the lights stay flashing until the phone is rebooted...

 

I have tried reducing the watched extensions to two on each phone and switched to TCP for the transport - but it has made no difference.

 

I have enabled full logging and with the direct line ringing you get a "notify" message going to all phones from the ringing extension with XML content showing the connection details. The same with when the call terminates. When you ring the hunt group I would assume I should get a message from each extension in the hunt group to all other extensions - but this doesn't seem to be happening (although the log gets pretty full so I can't say 100%), so I don't know if PBXNSIP is not sending out all the messages.

 

A hunt group is definitively a "stress test" to the phone. It receives a large number of messages in a short period of time. Maybe what you can do it log only SIP packets to/from the IP address of the phone and write the log to a file. Then we can find out if the messages are going out as they should.

[Buttons on 300's]

Hmm nope, should I? Just updated them to the latest version also.

 

Anyway... Just edit the template above and then you should be able to provision the 300 as you like.

[Buttons on 300's]

Just noticed that the 300's that have been auto provisioned on a new system have their "buttons" (mute, directory, redial etc) aren't working anymore, presumably as they're set to "buttons".

 

How do I stop the PBX doing that?

 

We addressed that in the next version 3.0. Short-term workaround is to put your own snom_300_fkeys.xml in the html directory:

 

<?xml version="1.0" encoding="utf-8"?>
<functionKeys>{if_button dnd none}
 <fkey idx="dnd" context="active" perm="RW">{enum_button dnd button+dnd private=line}</fkey>{fi_button dnd none}{if_button 1 none}
 <fkey idx="0" context="active" perm="RW">{enum_button 1 button+1 private=line}</fkey>{fi_button 1 none}{if_button 2 none}
 <fkey idx="1" context="active" perm="RW">{enum_button 2 button+2 private=line}</fkey>{fi_button 2 none}
 <fkey idx="2" context="active" perm="RW">{parameter key3}</fkey>
 <fkey idx="3" context="active" perm="RW">{parameter key4}</fkey>
 <fkey idx="4" context="active" perm="RW">{parameter key5}</fkey>
 <fkey idx="5" context="active" perm="RW">{parameter key6}</fkey>
</functionKeys>

 

Do you see the parameters key3..6 in the PnP parameters in the web interface (admin/pnp settings)?

[Address book]

Is there any way to get all of the existing extensions in a PBX into the addressbook automatically and keep it updated? Seems mad that I have to put them in manually?

 

For what phone type?

[Smon360 address book & language issue]

I am still having problems with the address book. It only displays the first 9 entries out of 20. I have even tried upgrading phones to 7.1.33

 

It have made a trace and it is the PBX that generates an incorrect XML reply to the phone request.

 

The PBX displays the first 9 entries that match. Because there could be thousands, there is a limit. The point is here that you need to "drill down" by entering more digits. For example, if you search "Valerio", you would enter <Adressbook>, 8 (TUV), 2 (ABC), 5 (JKL) than then usually the list should be short enough that you can use the arrow keys to select the right entry.

[Debian and Suse KDE Update]

I cannot get pbxnsip to work using KDE on Suse or Debian only Gnome. In fact Debian will only allow me to use pbxnsip using root not a user account even if I allow the executable to be run with root privellages. I have tried everything known to man kind and have come up with the conclusion that Gnome under SUSE or Debian is the only way to go but be sure to be logged in as root. If someone else has a suggestion on how to get the program to run in user mode then please write back.

 

We were thinking about an option so that after the PBX starts up it changes the account. That means after settings the priority and everything, the PBX would change the account to an user account, reducing the risk of malicious things and then start operation. Do you think that would help?

[Ports to forward, IPnet, FXO box]

I have 3 questions.

 

If they belong into different areas, it might be better to post them seperately. Makes it later easier to find the answers.

 

1. What ports do I need to forward through my firewall. I currently am forwarding SIP 5060 and 5061 as UDP and TCP

I changed the RTP ports to 16384 - 16484 and forwarded thoser too. and I have forwarded the SNMP 161.

I can log in via the web to my server with my public IP now but I can not register phones outside my LAN. My softphones say error 408 timeout when I try from outside my LAN.

 

Well, probably the easiest way is to check out the netstat command and find out what ports the PBX opened. The only dynamic ports (apart from DNS, which you don't have to take care of) is the RTP ports. A good read is http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses.

 

2. I have a Tadiran phone system and I would like to maker an IPnet in order to connect my PBXnSIP to my phone system. I would like to be able to use the lines on my system and be able to intercom back and forth.

 

Sorry I don't get the question here... You want to trunk into an existing (analog) PBX?

 

3. I have a Sipura FXO box and I want to know how to connect it.

 

Sipura sold their stuff a couple of years ago to Linksys, maybe http://wiki.pbxnsip.com/index.php/Linksys can help here.

[Passing Parameters to Snom Phones]

Well, that does not help because the file has no reference to the MAC.

 

Better do the following: Put the following file into the html directory and name it "snom_3xx.xml":

 

<?xml version="1.0" encoding="utf-8"?>
<setting-files>
 <file url="{https-url}/snom_3xx_phone-{mac}.xml?model={attribute model}" />
 <file url="{https-url}/snom_3xx_fkeys-{mac}.xml" />
 <file url="{http-url}/snom_general.xml" />
 <file url="{http-url snom}/web_lang.xml" />
 <file url="{http-url snom}/gui_lang.xml" />
</setting-files>

 

Then you can put another file into the html directory and call it "snom_general.xml", where you can put whatever you want to put in all of the 320/360/370 phones.

[Passing Parameters to Snom Phones]

I have tried coping the {MAC} directory under tftp and under http but it does not work!!!!

 

Copy a file from generated/{MAC}/file to tftp/file. The {MAC} should go away, so that the file is directly in the tftp directory.

[Spectralink 8002]

Well, we need at least one file with the MAC address in it. Otherwise it is really difficult to find out what account should be used for the device...

[Passing Parameters to Snom Phones]

We are using the PnP provisioning where possible for our clients. We have run into one problem so far with this. How can you configure parameters such as auto_dial through the PnP provisioning system? Normally I would use a Perl script to accomplish this through the web interface of the phones, but with the interfaces password protected, it is difficult to do this in any automated way.

 

Well, you can't do that through the web interface. But you can take the files out of the "generated/{MAC}" directory and put them into the "tftp" directory and edit them there.

[snom firmware 7.1.33]

can i update?

 

regards to this may not?

http://forum.snom.com/index.php?showtopic=71

 

IMHO it should be a clear improvement.

[Spectralink 8002]

Just got a pair of these today and they're "interesting" to setup. Got it working with PBXnSIP but the autoprovisioning is weird. Has anyone got these to work automatically?

 

Ehh - somehow the provisioning file got lost (we don't even find it any more...). Would be great if you can give us access to the file that you have ...

[wan port]

like i call my remote office instead of calling 999 ext 150 i just want to call 150 and get the remote party

 

As long as there is no overlap with an internal extension than that should be possible. For example, use the pattern "1xx" in the dialplan if you want to route 3-digit numbers starting with "1" to a specific trunk.

[Crackling and Popping noises]

I would try to upgrade to version 2933 (http://www.pbxnsip.com/cs410/update-2933.tgz). The version that you are using has a bug in the setting of the gain (it sets some other gain that actually can easily cause distortion).

 

Also, you should check the phones audio level again. Usually, users increase the volume of their phone when the gain is too low, and after changing the gain back to normal the phones are too lound and generate echo.

[Uninstall leaves pbxnsip-service active ?]

Probably nobody will ever want to uninstall pbxnsip, I did (sorry).

After uninstalling through 'Control Panel' I noticed that the pbxnsip-service was still present in the services-list, still marked as 'Start Automatic' and consequentially generating an error message in the event viewer after restarting.

 

Of course, a new install is not a problem and works fine. I assume this is a minor bug in de-installation, or is there something wrong with what I do ? (exept the de-installation itself ).

 

Using Version: 2.1.10.2474 (Win32)

 

Well, well. The service manager has its own ways... I remember uninstalling in 1.5 required not only one reboots, but a couple of them. Plus manually editing the registry. But I must say, I also don't have much experience uninstalling it . "In theory" it should work automatically by the installation tool.

 

But at least you can be sure it is not being started by accident and blocking ports 5060 & 5061.

[Moving Domains]

When attempting to move a domain with tel:alias configured from a server running version 2.1.6.2450 (Win32) to a server running version 2.1.10.2474 (Linux) the tel:alias did not move with the domain. The same issue occurs attempting the move from server to server of the same version 2.1.10.2474 (Linux). Any ideas?

 

Well, global names of course have a special meaning in restoring domains. If there is already another tel:alias with the same name, we are in trouble.

 

But it seems that all alias names are not restored. Looking into this right now.

[Set processor affinity to 1 failed]

after service reboot the log show this :

 

[2] 2008/06/06 18:20:50: Set processor affinity to 1 failed

 

Does Anybody know the meaning of that?

 

This is only important in environments with multiple CPU cores. If you have several cores, it means that the OS might shift the PBX process around between the CPUs - every shift meaning that during that time, there is no RTP. We had cases where this meant a lot of jitter and playout problems.

 

I assume you are using Linux?

[did number]

i would like to know the advanteges of adding a did number to the fxo ports

 

The presence of a DID number tells the FXO driver that this line should be tried for outgoing calls. And you can also use the DID number to perform inbound routing based on the DID.

 

and secondly why does the pbxnsip not detect that the user has hanged when it goes to the voice mail

 

i mean it keeps the line busy until i dont restart the pbx(not logical)

 

please dont tell me the the line is the problem as the line is hardly 1 metre from the central building exchange

 

That problem should be solved by settings the busy tone detection. I think there some other posts on this topic - is it stipp open?

[wan port]

is it possible

 

Yes. That is actually the reason why there are two ports.

 

In contrast to other products in the market, the PBX can deal with many IP addresses. You don't have to worry about the RTP flow and the address translations.

 

One common problem is the default gateway - you must make sure that you have only one. When you get the IP address by DHCP on LAN and WAN, then usually you have two default gateways and that screws things up in Linux. Therefore on the latest and greatest we put some JavaScript in the web interface that presents a warning in the dangerous cases.

[did number]

for kuwait the dialtone is continous so how do i put that

 

Good question... I think that tone 1 and tone 2 apply to the busy tone, and that the dial tone is always continuous (we need to verify that). That is because there is a seperate flag for dialtone.

 

I would suggest turning only busy tone detection on, set the tone 1 on/off correctly and give it a try. Then if that does not work turn dial tone detection on as well.