Jump to content

Vodia PBX

Administrators
 View Profile  See their activity

  • Posts

    11,110

  • Joined

  • Last visited

 Content Type 

Posts posted by Vodia PBX

  1. Call Pickup

    in snom Phones

    Posted

    We have groups of people who are able to pickup calls within their pickup-group. This is done with the buton feature from snom/pbxnsip. When someone pickup a call from another an increasing originator-numer is shown.

     

    1. pickup "*6015775"

    next "*6015781"

    next "*6015784"

    ....

     

    On the called phone the right originator is shown. Do I have something to configure?

     

    That translates into *601 meaning pick a call up and 5775 being the call-identifier.

     

    The PBX automatically tells the phone what code to use for the next pickup. So you don't have to configure anything.

  2. No VM

    in Mailbox

    Posted

    SDP looks fine - setup is Vega ISDN->asterisk->PBXnSIP. It's been working fine until (as far as I can tell) we installed the v3 build, though that could just be co-incidence. Also all normal incoming calls are fine, it's just when it tries to send it to VM.

    I can PM you the pcap if you want? It's only 160k

     

    Well in that trace we see that the Asterisk drops the connection.

     

    When the PBX picks the call up, Asterisk tries to re-negotiate the codecs (obviously because it wants to change the IP address of the RTP) and then sends a BYE (no idea why). Maybe you have to upgrade to the latest 1.4 version. Maybe you have to put something like canreinvite=false or something like that into the configuration files, so that it does not attempt to make the PBX talk directly to the Vegastream.

  3. Hunt group extensions call fork

    in Hunt Group Setup

    Posted

    Is there any way to allow the extensions that are ringing via a huntgroup to still call fork to a cell phone?

     

    There was another post recently.

     

    Bottom line. There is (static registrations). However, be prepared for a lot of problems with offline cell phones that will pickup immediately.

  6. VM-to-Email Email Timestamp Bug in Version 2.1.12.2489 (Linux)

    in Mailbox

    Posted

    The bug is in the timestamp of the "Sent:" field of the VM-to-Email Email you receive from Pbx-n-Sip, the time is set to GMT (Greenwich Mean Time) and ignores the "General: Timezone:" setting on the Admin System Settings page.

     

    The timestamp accessed through the web interface or through the phone itself is correct and unaffected. And, of course, you have your mail program's report on when the email was received, but seeing the email come from 5 hours in the future without annotation of GMT makes you do a double-take. (I'm in Central Time Zone.) I would rather it just show the correct time.

     

    The email timestamps are always in GMT. That is because the one who receives it might not be in central time. The send time is corrected with an additional argument which looks like -0800 meaning subtract 8 hours. There was a problem with the second argument, but that should be fixed now in the 2.1.12 version.

     

    Look for a header like this: Date: Tue, 22 Jul 2008 04:45:55 +0200

     

    Also, why would this be handled with no problems in earlier versions but somehow become an issue an more "advanced version" -- I thought this fundamental and "old hat" part of things was held down tight already.... or when you make advances you "blow up the code" (including things that WORKED normally) in order to do so? :o

     

    You make one step forward and count the steps backward... Welcome to the software world :o !

  9. ACD Calls going to ext do not redirect

    in Agent Group Setup

    Posted

    We have a rep who is rarely at his desk, therefore he has his ext set to redirect immediately to his cell phone. However, when he is logged into the ACD queue the calls coming into the queue do not redirect to his cell phone. Is that normal? Is that a bug? Is there an easy way to allow the user to login to his phone and have the ACD calls redirect to his cell?

     

    No, that is not a bug. The idea that people fork potentially calls to a lot of cell phones (half of them being powered of, in the tunnel and redirected to the mailbox) just do not sound like a solution to me. The agent "mailbox" always picks up immediately, and the PBX has a very hard time figuring out if it is a natural person picking up or a machine.

     

    Though there is a work-around. You can add a static registration to the extension, then the PBX will also include the cell phone in the group. Maybe you can try this and see if it meets your business demands.

     

    I believe it is better to try the agents in house (potentially using hot desking), then if that fails escalate the call to a specific extension and potentially also forking it to a cell phone. If then the mailbox picks up, okay then it is last resort and even the caller cannot expect much more.

  10. Call Forking Timer

    in Extension Setup

    Posted

    With the latest fix to caller ID with respect to attended transfers, is it possible to get an option to set the call forking timer for the extension to 2 seconds. This is a good compromise for people who would like immediately, however will be getting some of their calls via attended transfer. This 2 second timer would allow the transfer to be completed in just enough time to show the correct caller ID on the cellphone.

     

    No problem!

  11. Call features from Cell Phone

    in Extension Setup

    Posted

    If a call is forked to a cell phone, is it possible to use features such as call forward via star codes?

     

    Unfortunately, that is not possible yet. The way the PBX does it is by putting a call on hold first, and that is a feature not supported very well from the cell phones (or the base stations). The PBX just hears MoH.

  13. M3 phone & one way audio

    in snom Phones

    Posted

    Just got a M3 set up on a pbx call center 25 and I get one way audio for 30 seconds...always 30 seconds, I have it supporting 2 registerations, one by itself and one with another phone..both registerations will work the same way...It is not always one way, it connects about 33 to 25 % of the time when answered or called.. Not much in the GUI to change on the M3, currently have it registeded to the Pubic Ip Address on the server, but it is on the same lan as the server..?

     

    What versions of the PBX? What version of the phone?

  15. Mobile Phones That Support SRTP andTLS

    in General Setup

    Posted

    Anyone have any clue on what Mobile Phones use SRTP/TLS so that I can connect to the pbx with security?

     

    Counterpath had a version that runs on Windows Mobile, but I have not seen a publically released version.

     

    The alternative is to use VPN, which is supported on the Nokia WiFi-enabled phones. It does not use SRTP, but VPN makes sure the call is being encrypted.

  16. Call-back

    in Product/Site Comments

    Posted

    Do You know when You will relase version 3?

     

    It would be good to have callback function.

     

    It is still not done, but at least it seems we now know that it will be done using the calling-card account. There we can use the authentication mechanisms and then we can do the call-back. Authentication can also be done while the call is still not connected, so that the caller will have no cost.

  17. License woes :(

    in General Setup

    Posted

    Well I bought a copy of the software and can't seem to get my license key issues sorted out.

     

    First the mac address was typed in wrong by a pbxnsip employee and now several keys and requests for assistance have been unsuccessful.

     

    Has anyone else had issues with their key? I'm having a difficult time recommending this product to my clients at this point! :)

     

    Understandable. It is frustrating to waste time on something like this.

     

    Because of that, I always recommend to use copy & paste even for short strings like the MAC address... It just makes this much more i*-proof..

     

    The alternative is to use a USB dongle. That also has the benefit of making the license independent from the MAC.

  18. click to call link in voiceamail email

    in General Setup

    Posted

    The email we receive with a voicemail has a nice feature that is a link that will initiate a call back to the caller.

    It is just a click to call url (nice option available) however, this link does not work for us for the followng reason:

     

    The links contains the public Ip mapped in the "IP Routing List" field. Which is good in some cases but It should contain the domain name instead. Many routers/firewall/nat won't allow connections to its own public from the LAN.

     

    There is a settings called "c2d_root" that you can set (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File). If you specify the DNS name there, the PBX will be more than happy to use the string you provide there. For example, you can put there "http://pbxnsip.abptech.com".

  25. Registering ITSP accounts on the CS410

    in ITSP's

    Posted

    That makes sense and i have tried it but i am still getting the same result, (one way audio on inbound call, outbound is ok) and i have now tried 2 different NAT routers. Is there any types/models of routers that you are aware of that can handle this type of routing?

    Do you have a recommended setup for situations like mine?

     

    Of course IPv6 or at least a public IPv4 address. It all comes down to the question if the PBX can present a routable IP address and port. If you don't have a routable address, you can not seriously run a PBX there...

     

    Those port forwarding and DMZ games are really extremly support unfriendly. Sooner or later, you'll get the next issue because some idiot in the network blocks the RTP ports on the DMZ or something else. Finding those kind of problems takes extremly long time, and in the eyes of the customers it makes you look goofy. It is simply a fix, not a solution.

     

    Also what is the reason for provisioning phones on the WAN port and not on the LAN?

     

    In principle the device has two Ethernet interfaces. They are called "LAN" and "WAN" just because marketing read somewhere that devices must have two ports (us technical guys, we could of course also configure two IP addresses on the same port or just use VLAN, but that is hard to imaging for people trying to sell a tangible product). Is the LAN connected? There is a problem if you have two IP gateways on the system, then the Linux gets into trouble. The 2933 image has a JavaScript trying to protect the admin from entering such dangerous combinations.

×
×
  • Create New...