Vodia PBX
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what string would i put in to make a static registration for my cell phone?
If you have a local PSTN gateway, you would use something like "<sip:12121234567@ip-of-the-gateway;user=phone>".
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Actually it's not a cellphone. It's a receptionist who will answer this phone, appearantly via POTS.
If I don't fill in a cellphone but a normal phonenumber, the receiver hears a tone/message?
Of course the PBX does not care if it is POTS, cell phone or just a SIP URI (actually, there is no way of finding out). The point behind the cell phone inclusion in an extension is that the caller actually has no chance to find out that the call was redirected. The PBX user should be able to hide where he actually is. That is different from the redirection after timeout, when the PBX makes an annoucement ("please stand by while redirecting") and then actually plays the ringback tone of the POTS line (which may include comfort noise in the beginning).
The biggest problem is the propagation of the caller-ID. It is a reality that most ITSP are today not able to differentiate between display-ID and network-ID, and then those redirected calls always seem to come from inside the PBX. Playing an annoucement could be a "poor mans" original caller-ID - and it would also solve the problem that cell phones tend to redirect calls to mailboxes.
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That would affect the ability to properly PNP, wouldn't it?
Well, only the dialplan part... As VoIP becomes more mainstream and people have cell phones, hitting the "call" button on a SIP phone also becomes more popular these days.
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Never had much use for TAPI, nor did many other people considering its reputation. But people like to see the fancy stuff, knowing they'll never use it, so why not go ahead and get TAPI working if for no other reason to show off. Follow the ever so simple WIKI DOCS and on my first HOME PC setting behind a good firewall set to allow all internal going out, the results are less than spectacular. The PBX is setting on a Public IP address and the results are "Internal Error Close and retry" Darn.
Will retry on the Local LAN on another PC with a local IP connection to the PBX.
But in looking for how to diagnose the TAPI stuff, I came along the following TAPI related items
I thought I might share this, and hopefully someone with successful TAPI installations might make some benchmarks or tests with these tools and report back some tips, I'm certain to.
Yea, the TSP is using plain SIP, unencrypted and using UDP. If you have a SIP-aware firewall between the PBX and the PC, trouble is pre-programmed...
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When we calling with the tapi driver the connection is broken after between 30 sec and 2 minutes.
When just dialing the number directly on the phone everything is fine.
"Broken" means the PBX disconnects the call? Do you see that the PBX sends a BYE? Is there NAT between the PC that runs the TSP and the PBX?
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The part I'm getting tripped up on is: How do I accept calls from the internet for that speech server?
One simple workaround would be to use a manual registration (see http://wiki.pbxnsip.com/index.php/Manual_Registration, and http://wiki.pbxnsip.com/index.php/Extension#Registrations). Then the speech server does not have to register with the PBX, but you can treat the server like a registered SIP device.
From what I've read, I should be able to create a second trunk (a "SIP Gateway") called "localhost", with "localhost" in the "Domain" and "Outbound Proxy" fields and with "Global" set to "yes" and that trunk will handle all incoming traffic that isn't detected as a local extension. Here is the line from the 2.0.1 documentation that makes me think this should work this way:"The domain name "localhost" matches any domain name presented in the Request-URI, as usual." - top of page 108
It doesn't seem to matter how I twiddle the settings, I keep getting:
[8] Could not find a trunk (2 trunks)
[5] Received incoming call without trunk information and user has not been found
While I was experimenting I put the IP address of the client user agent into the trunk's "Outbound Proxy" field and that changed the messages to:
[5] Identify trunk (IP address/port match) 3
[5] Trunk localhost sends call to 7000
[5] Trunk call: Could not identify user
I don't know what "Could not identify user" means. The user who is making the call? The user who is being called (7000)? Something else entirely?
The "outbound proxy" is also the "inbound proxy" - at least the the sake of finding out where the call came from.
The message "Could not identify user" means that the use was not found in the domain. Is 7000 an extension or account? Maybe you should create 7000 and add a manual registration pointing to the speech server.
In case it matters, I'm using a demo license at the moment until I know for sure I can get this working.Demo license makes no difference.
Also important is that you check "accept redirect" in the trunk to the speech server. The speech server is nothing else that the SIP engine of the Exchange UM. http://wiki.pbxnsip.com/index.php/Microsoft_Exchange should contain a good checklist on how to configure the speech server as well.
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Yes I tried that. However we are using Exchange 2003 not 2007 so UM is not even an option., I was just trying to get it to send a cdr report to my email or send me a voice mail. I found that if I removed the username and password that it would send email just fine.
Oh ok, SMTP (thought it would be SIP) - if you do IP-Address based authentication, then you don't need username/password.
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Sometimes we need to forward some SIP trunks to another telephonenumber temporarily but we would like the receiving party receives a message or tone he/she knows it's a forwarded call.
For e.g. when you call +31201234567 you would hear as a caller "one moment please your call is being forward" and the other party to which the call is forwarded receives first a message 'this call is being forwarded from office A' and then forwards that call.
Other solutions are welcome to, we want to hear something when a call comes from this forwarding.
On level of extensions or trunk, both solutions are welcome.
Hmm, what is the use case here? Is it only for the cell phone? There we still have the topic that we want the user to confirm the call because of the cell phone mailbox problem.
It would also mean that the first few seconds of the conversation are cut off? Or do you want to mix it with the audio of the other side?
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http://provisioning.snom.com/update6to7/update_once.php
I found this in snom_3xx_phone.xml ,
where can I change this ?
Report the MAC to snom. Maybe there is something specific with that MAC address.
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Hello again. I am trying to get the simple exchange integration going with Exchange 2003. We have SP2 installed. When I attempt to connect to Exchange from the PBXnSIP server I get an authentication error 504 unrecognized authentication type. Can anyone telll me what is going on? I have logged into the webmail account I created for the PBX so I know the username and passwords are good. Thank you all for your help.
I guess you checked http://wiki.pbxnsip.com/index.php/Microsoft_Exchange?
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snom 360
You should really use PnP, at least for the initial config. The address book requires that you calculate a hash over the username and the password, and that is really difficult if you want to do it one your own.
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Hi i am having an issue where the phones will only show the addresses of the extensions of the pbx. the client needs his address book...can someone please provide info on what string i need to put into the phones/pbx for this to work?
What phone model?
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I would factory reset the phones and then if it still not works power-cycle the switches. Also, make sure there is no VLAN involved here and the netmask is set correctly in the DHCP server.
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Hey two weeks later and still having the same problem. i am setup as follows.
cs410 with pnp to snom 360s. the inbound trunk rings to a hunt group for 15 seconds before hitting the auto attendant.
if an employee picks up the phone from the hunt group (in the first 15seconds) that call cannot be monitored by the other phones?
if i switch the trunk to ring to an agent group for the first 15 seconds i do not have this problem!!
please advise as i promised my client call monitoring and he is driving me nuts!!
Are you using buttons or dialog-state? I tried this here with buttons and it seemes okay to me.
Try hitting the save button on the button profile again. If you change (or create) an account that you list there after hitting the save button, references might be broken (we'll fix that problem in 3.1 as it really can drive people nuts).
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Just to make sure we are talking about the same kind of issues here:
The software is running on a Teles Box, we are not using TLS and are using udp, the phone we use are Snom 320/360.
The main issue is that when a second call comes in the processor load briefly goes to a 100% and at that time the audio drops real shortly.
Immediately after that the load becomes normal and the audio is flowing again.
Hmm. Hmm. Hmm. But you do see "Set scheduling priority to ..." on log level 5 after a start? That should tell the OS that RTP is more important than anything else. I remember there were some patches in the kernel with the scheduler. Maybe they had the side effect that the RR scheduling does not work properly any more. We have to check.
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It's easy to make a Dial Plan for "Dial Extension" and assign the Lobby Phone to that Dial Plan, but to give that Extension E911 access, it seems another Dial Plan with access to a Trunk with 911 dialed sends 911 followed by an ERE expression to strip all but the first three digits is required.
Is this how others have done this, or can the main dial plan detect the restricted extension and do the work?
I would use: Pattern: x11, no replacement. That should already do the work.
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OK. That corrected the problem. Now when is this going to become GA?
:nailbiting: Maybe this week.
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I have a client who wants to keep their standard fax line in each of their offices, but then have broadband added to it. Has any one had any problems with this,, i.e getting interferance from the fax machine when sending or transmitting...
Maybe http://www.pbxnsip.com/download/qos.ppt is a interesting PowerPoint for you. Make sure that the broadband is reliable to transport voice, that will keep trouble away and make the customer happy.
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I need to accept 411 from the dial plan and pass *411 to the carrier.
how do i do that? * is a wildcard do i do \* or something?
Yes, use "sip:\*411@\r;user=phone" as replacement.
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Is their a way to change the pin number on a conference room, without actualy giving the customer access to the pbxnsip domain level web interface..
No. The user can only set up scheduled conferences where the PIN code is specifically assigned for a specific user.
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I am adding <volume voice.volume.persist.handset="1"
to all the <macaddress>.xml files in the generalted file for the Polycom phones , obviously alot of work , how can I do this for all phones without editing every file , shouldn't the there be a default file, like 000000000000.cfg file somewhere , or a master XML ?
Well, we have two options: Make is accesible through the PnP settings of the PBX or always set it to one. Votes?
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I had to go to every polycom_phone.xml file and change the "1" to "0" to disable missed calls tracking , is there a better way to do this ? the hunt group calls fill the display and on Polycom 320/330 phones it's like 10 keystrokes to clear the missed call list , since they are on a hunt group it constantly clutters the screen with the missed calls
Well for that problem RFC 3326 has been invented. However, Polycom seems to support a proprietary feature called "calllist-missed". However, I quickly searched and found nothing on the topic.
Maybe we just turn the local missed call lists off. I agree, having the messager that you have 323 missed calls is not very useful.
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Do you have a Linux version? I have a test server that I will try it on and it is Linux.
What kind of Linux flavor do you prefer?
how do I connect a spa400 to a trunk in ver 3
in Gateways
Posted
The web site says "Designed only to be deployed with the Linksys SPA9000". Not sure how serious this is. Maybe it is only supporting Skinny. But if it works like the other SPA gateways it should be pretty straightforward. On the PBX side, you just need to set up a gateway trunk. In FXO, you don't have the chance to do too much with caller-ID anyway, so the trunk setting default should be fine already.