Vodia PBX
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example: We have legal rights to 4 telephone numbers. Some are business numbers some are personal. We have a SIP trunk service.
If I am making a business call I want the call to go out with the FROM (Callerid) being the business number. If I make a personal call I want the callerid to go out using the personal number.
How do I make pbxnsip set the apporpriate callerid based upon the call I am making from 1 extension?
For this, you should use the ANI field. You can set it in the extension, and anything else that is able to make a phone call. If you want to use the same ANI for all extensions, you can also set it from the trunk.
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Now I have the half solution found myself. There is a totally different behaviour between setting up a Park Orbit Button in the Web-Interface of the SNOM Phone and setting up a Park Orbit Button in the Button-Settings of PBXnSIP. The last does something magic: Pressing the button while a call is active, it dials *85EXT (where Extension is the specified Park Orbit Extension in the Button Settings of PBXnSIP). Pressing this button while a call is in the Park Orbit causes the phone to dial *60RANDOM (where RANDOM is a irreproducable Number) I absolutely cannot understand, but that work.
BUT: Now I still have an unresolved problem.
This only works as long as calls received on SIP-Identity #1 of the SNOM Phone. On all other Identities (same PBX of course) I doesn't work. I always get the voice-message that the call couldn't be parked.
Multiple identies on one handset are always a headache. If you are using buttons, you can tell the phone what identity context to use for a button. The PBX usually stays out of this dangerous water; but in your case it might make sense to bind it to a specific identity.
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This is v4.1.12.0037 boottom and v3.1.1 SIP
Okay, this is what I did: I nuked the phone several times with a factory reset, and settings reset. After that, set it up for PnP with HTTP, and username/password set appropriately. It works.
I did not try to downgrade the bootloader. Maybe the older bootloader has the problem that authentication is not supported yet.
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So does it already go to the IVR node?
Try using the auto attendant instead of the IVR node. Depending on your DTMF setup, the IVR node might not be able to detect the CNG tone. As you are going to the AA anyway, that should be no big change. Then you can use the "F" in the direct destinations.
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Callcentric.com is now offering this:
Pass Caller ID in SIP INVITE message
This feature primarily applies to business customers and customers running their own IP PBX system. It will allow you to pass Caller ID as any phone number on your account or any external number that has been verified by Callcentric within the SIP INVITE message of an outbound call.
Send the Caller ID number of a DID on your account or an already verified number within the SIP INVITE message of an outbound call. To do so, have your IP PBX attach any of the following headers which are supported by most IP PBX's (check your vendor's documentation for support) to your outbound calls:
P-ASSERTED-IDENTITY
P-PREFERRED-IDENTITY
REMOTE-PARTY-ID
For more: http://www.callcentric.com/new/#caller_id_in_invite
Now with this new service available what would we do in PBXNSIP to use it?
Callcentric says "(check your vendor's documentation for support)"
We added a new header called "Related-Call-ID" that makes it possible for the service provider to check if the Caller-ID is a valid redirection or the caller tries to spoof a caller-ID. AFAIK CallCentric supports this header. A quick try will show this.
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When you schedule a conference, can the system send a call to the participant and if they pick up they can call the extension of the conference ext #.
Well, we had some code in there already when we realized what the real problem will be - in most cases there will be a mailbox picking up or just no answer or busy. Then the next question is if the PBX should give up inviting that person or retry after a time.
If it is about saving the participant the cost for the call a transfer into the conference is the better answer.
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I have a potential client that would like to install a telephone switch to replace its many PSTN lines. They have modems on their PC's that currently dial numbers for them from a legacy application that talks to the modem through COM2. Is there a way that I can redirect these requests to COM2: to the PBXnSIP TAPI client - this would let me get the phones in place without them needing to update their application?
Maybe it is easier to use the click2dial feature that just uses a HTTP request. Maybe you can just start the web browser from the command line and provide it with the correct link to start the dial.
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I believe this is because the PBX does not send RTCP. Seems like we finally have a reason to put this in.
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It would be a lot easier to manage domains including backup and moving a domain to a different server if each domain had a separate folder with all its files including the audio files. Backup or copy the folder and you would be done. The existing export will move the control information but not the audio files. It would be nice if the log file was visible withing each domain and it would show only entries for that domain. I would think having a separate instance of PBX would be a lot safer and eliminate any problems between domains.
I agree on the domain part. It would make life easier for moving domains around. The reason for the flat architecture is that relational databases use a integer as index. We would have to change that into a domain-context index. Not sure how much that would be a problem, but it is worth thinking about it.
The logging is more difficult, as many log evens simply do not apply to a domain. Also IMHO the domain admin should not dig in the log; this is the job of the system admin.
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A nice feature would be a substitution list that would display a different Caller ID name on incoming calls. Frequently the name portion of Caller ID is missing or wrong. This list could be for the entire domain. One extension wouldn't need a different name than the other extensions.
Can we somehow get the address book do this?
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I am in Costa Rica. I am using the only carrier available (state monopoly). I can connect my Panasonic phone (bought on Amazon) directly to the phone jack, and I get caller id info.
You should change the PSTN gateway logging flag (admin->logging settings), set the log level fairly high (e.g. 9) and reboot the device. Then make an inbound call and check the log for information on what the gateway sees regarding caller-ID. Maybe Costa Rica is the next country that had it's own idea on how a caller-ID should be presented...
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im running a little older version so i put this snom_3xx_custom.xml in my html directory.. and it doesnt seem to update.
You should see in the log file of the phone what file it is trying to fetch.
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Is PBXnSIP planning on adding stronger logging mechanisms. I'm looking to determine the missed logon attempts from devices trying to register to our managed servers. As it stands, there are no entries with any specifics I can refer to.
If someone tries to unsuccessfully log into any PBXnSIP devices, my only mechanism of knowing is having the client tell me they can't log in. While I can look at *some* of the SIP information on the phones, there is nothing on the PBXnSIP boxes themselves that tell me what the issue is, e.g. incorrect password, unauthorized login, etc. I cannot tell who is attempting to log in, IF they're getting any errors, if they're even authorized. Is there a mechanism to get at least syslog entries up and running.
3.3.1 is writing a syslog message when a admin logs in, including the IP address. This also goes into the web interface. Unfortunately, because syslog is crazy difficult in Windows, this works only in Linux.
Brute-force attacks are slowed down significantly because an unsuccessful login takes several seconds and the PBX lets other requests wait before it accepts them.
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"Waiting Time"
"Ring Time"
"Talk Time"
"(Hold Time)"
The waiting time is the duration the caller spent listening to music and annoucement (no agent taking care about this call yet). The ring time is the duration while the caller heared ringback tone (agent's phone should be ringing here). The talk time is the duration the call was connected, including the duration the agent put this call on hold. The hold time explicity shows how long the call was on hold, the brackets around it indicate that this is part of the talk time.
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With the office edtion of pbxnsip, IS there a concurrent limit?
I thot there wasn't one (of course really there is when cpu load too high) but is there a hard limit?
Hmm. Really good question (I don't know). You can always check by base64-decoding the license key. If you see something with "call" and a number behind it, then that's the limit.
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I have a client with 1 voip trunk, (2) cable voip lines on a Patton SN4114 ports 1&2, (2) analog PSTN lines on ports 3&4. They only use the analog lines for incoming and DR of the voip trunk(s) - no outbound calls unless everything else goes down.
Ideally they would like calls to go out the cable voip lines first, then if those are filled use the voip trunk. If that is down then go back to the patton and use the pstn lines.
Is this possible using the same patton gateway? I currently have a hunt group setup to use just ports 1&2 for outbound. Incoming on all 4 ports works just fine.
You could use the failover feature of the trunk to let the PBX try one trunk first and then move down the dial plan on another trunk if the first trunk fails. Having only a certain number of CO-lines will also help to limit the number of calls on a trunk; you might not need it if the gateway sends a 5xx code when all lines are in use already.
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How do I setup a day/night service flag as a button on a 650 w/ sidecar? And on a 550. Ideally they would like it to light red on night mode and off in day mode.
AFAIK Polycom has no way of explicitly turning a LED on or off. The PBX has to pretend that there is a call going on on the resource that they subscribe to ("BLF"). Did you try to monitor the flag the way you monitor an extension?
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Is there a limit to the number of participants in a conference room?
Is it linked to the number of concurrent calls?
Yes, the concurrent calls are one limit. The CPU load should also make sure that there is not too much jitter coming up.
If you want to keep the participant circle closed, you should invite the participants. Then only users that know the PIN can join the conference.
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I got the message, that I could put some files into to html OR tftp folder, and then I could update the phones with the new PnP settings; BUT I need to make a file for each phone; And I think this will take the same time, as going online with a customer and correct their phones.
We made a fix that allows the admin to manually put files into a folder called "provisioning". That makes it possible to upgrade phones in the field. Obviously we want to push the responsibility for publically exposing passwords to the admin. So keep these files there not for too long.
It also makes clear why we needed to do this. It is not that we got bored; there was simply a security hole that we needed to close. Unfortunately, it is not possible to do this fully backward compatible, as the phones out there simply are not able to authenticate themselves.
The attachment can serve as template (however - untested, so a feedback would be great).
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I've just installed 3.3.0.3165 on a windows 2003 server and I'm having several issues with provisioning my Aastra phones. The phones seem to not pull anythinig down when trying to use http and tftp pulls down the base configuration but nothing specific to the exension. I do have windows firewall running but I've added the pbxctrl.exe to the allow list. I was previously using 3.2 on Linux and I was having no trouble getting the phones to work and the log files are showing nothing useful and nothing is being written to the Generated folder.
Should be fixed in the 3.3.1 release.
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It was our mistake: there was a firewall after all and it was rejecting UDP packets coming to high ports. What's strange now is why on earth some routers had no problem sending audio to the server although UDP traffic was blocked!!!
Whow. Never stop learning!
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Yeah I would really like to see some sort of fix. I can even leave a message and hit 0 and I get the sound at the end of the message as well.
This problem might go away with another fix that we needed to properly record DTMF during a conversation.
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You have to use DTMF for this unless you want to record all calls. AFAIK Aastra does not support a magic record button that works without DTMF.
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We will provide the possibility to provision the web client passwords. For this, you will have to put files into the html or tftp folder, this way the devices out there will get the password. This is again "safeby by MAC", but at least once the upgrade is over you can remove the files and continue normal operation.
For new devices, you should make sure that the devices have the username/password already in the device and the customer knows how to put it into the phone after a factory reset.
Call record using snom phones
in Mailbox
Posted
The setting is called "send_recording" and it is a global setting (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to change it).