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Vodia PBX

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  4. Setting Up Seond Domain

    in General Setup

    Posted

    If I am understanding you correctly, we should;

     

    Setup a DNS name externally and internally called "pbx.domain.com"?

    Setup an additional domain in PBXnSIP called the same.

    Point DNS for this host at the PBX servers IP.

    Try to have the phone authenticate using 350@pbx.domain.com and password?

     

    I'm sorry I've not used anything but the local host domain before.

     

    It is a significant jump from one domain to multiple domains. You need to be more precise with the naming.

     

    The important part is on the phone:

     

    Outbound proxy: It is not really neccessary to set up DNS for that, you can use the IP address. You can also put the same outbound proxy in different domains, e.g. "pbxserver1.hosting-company.com". But of course, it is much more flexible to use real DNS names for the outbound proxy, for example "client1.hosting-company.com" or "pbx.client1.com". Maybe it is time to look into DNS SRV records.

     

    Domain: There you have to be strict. The domain must string-match the domain name on the PBX. It is the only chance for the PBX to find out where the request should be sent to.

     

    My suggestion is to use the outbound proxy all the time. It makes features like call a missed call from the phone much easier, the phone will not try to bypass the PBX and go direct.

     

    Also, you should not use the name "localhost" any more (unless you want to "catch" requests that are not going to any of the local domains).

  5. multiple interfaces

    in Linux-related Topics

    Posted

    Does anyone know how to configure pbxnsip with multiple interfaces? The application is running under Linus Redhat.

     

    You don't have to configure anything, just make sure that the routing on the host is okay. The PBX can deal with any number of interfaces.

  9. Distinctive Ringing

    in Extension Setup

    Posted

    I think the easiest is to just try it out. You can just edit the ringtones.xml file and set the Alert-Info header as you like. All you need it to edit the file and load it through the web interface (admin/settings/configuration at the bottom).

     

    <?xml version="1.0"?>
<ringtones>
<tone name="custom1">
<vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor>
</tone>
<tone name="custom2">
<vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor>
</tone>
<tone name="custom3">
<vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor>
</tone>
<tone name="custom4">
<vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor>
</tone>
<tone name="internal" type="internal">
<vendor type="alert-info"><{to-uri}></vendor>
</tone>
<tone name="external" type="external">
<vendor type="alert-info"><sip:1234567@test.com></vendor>
</tone>
<tone name="intercom" type="intercom">
<vendor type="call-info"><{from-uri}>;answer-after=0</vendor>
</tone>
</ringtones>

     

    Try an internal or external call and you should see the Alert-Info header set in the INVITE. You can modufy the ringtones.xml file on your own and see how the INVITE changes. Maybe we don't haveto change anything in the PBX to support this feature.

  10. Distinctive Ringing

    in Extension Setup

    Posted

    Okay, but then we are talking about the called number indication. The time between the first ring and the second ring on FXO is used to send additional information, e.g. the time/date, the caller-ID and it can also indicate the called-party (see DDN, for example http://www.nmscommunications.com/manuals/6709-13/appc.htm).

     

    That information is in SIP carried usually in the To-header, not in the Alert-Info header. If you are using a ATA, check the options to send the called number on the FXS using the MDMF format.

  11. Cellphone Access

    in General Setup

    Posted

    Is there a way to Release the first call and make a Second call easier?

     

    Not yet.

     

    How would you tell the other side to disconnect the call? The only chance is to wait until the other side hangs up, then we could go back to the first menu.

  12. Setting Up Seond Domain

    in General Setup

    Posted

    I have not before setup a second domain but am now finding that we need to to segment some other user's with their own billing trunk and logical grouping. I've have the domain setup, the trunk working but am uncertain how to get a POlycom phone to connect to this new domain. I can connect to localhost without a problem. Any help would be appreciated.

     

    There are two important settings. One is the domain name and the other one the outbound proxy. For domain identification, the PBX uses the domain name. The phone uses the outbound proxy for routing purposes. If you keep these two things seperate, it should work fine.

     

    And if you are using plug and play, it should also work fine.

  13. Extension Conference

    in General Setup

    Posted

    I can create a 3 Party Conference from my phone (2 outside parties and myself) by using the COnference button.

     

    I can not add any more parties to my conference.

     

    Pressing the Conference button produces no action.

     

    Is it possible to have more people in the conference using the COnference button?

     

    The 3-party is today a mainstream feature of the SIP phone. There are some phones that are able to have more than 3 parties in a conference (for example, the snom 200 was able to do that). Then this is no problem.

     

    If you want to transfer the participants into a PBX conference, you must blind transfer the participants into the conference room. If there is no PIN everything is easy. However, if you want to use a PIN things get complicated. AFAIK Polycom supports a button that blind transfers all calls into a conference room, that seems to be the mainstream way of moving a conference to the PBX. The big question is authentication and overlap avoidance. You don't want to accidentially bump into another conference held by another extension.

  14. Distinctive Ringing

    in Extension Setup

    Posted

    Is there a way to produce distinctive ringing from a unique called party number? Distinctive Ringing is a rather common telco feature that allows the user to place a fax machine (or other device that can detect DR) on a shared extension with a phone.

     

    The IAD that we use supports it if the PBX sends the appropriate code. Do you support multiple distinctive ringing patterns in the Alert-Info Header in the Invite message? If so, how is this invoked? Please be as specific as possible.

     

    The PBX already supports distinctive ringing, for example in a hunt group you can select that DR you want to use. There is a XML file ringtones.xml (see http://forum.pbxnsip.com/index.php?showtop...ed&pid=2246).

     

    I don't understand how you want to use DR to make the FAX selection. I also don't understand why you want to use the same extension number both for FAX and voice.

  15. Moving from VM to VM

    in Mailbox

    Posted

    In our previous phone system after leaving a message for a user you could hit *T to transfer to another ext, this way if you called in you could leave a message for someone and still try to get to another user without having to call back. I can't find this type of feature, is it available?

     

    What you can do is to define the "mailbox escape account" on either domain or extension level. If you point it to an auto attendant, callers can go there after leaving a VM message (after pressing #). Then they can dial whatever they like.

  17. No coded available for sending

    in General Setup

    Posted

    [8] 20080513162018: No codec available for sending

    [8] 20080513162019: Last message repeated 213 times

     

    What is this exactly?

     

    That means the PBX is supposed to send out RTP, but does not know (yet) what codec to use. It can happen when the codec negotiation takes longer. If it still appears when the call is already connected then it is a sign that something went wrong with the codec negotiation.

     

    We took this message out in later versions, it is not really neccessary and it cost a lot of performance.

  18. sip trunk on lan between appliances

    in Trunk Setup

    Posted

    I was wanting to set up a trunk on my Lan using private IP's from a cs410 through a switch to another cs410, with out going onto the Wan, is this possible?

     

    Only if you register the trunk to the other PBX. Then from the PBX on public IP's perspective, that registration is just a regular extension.

  20. BHCC and BHCA

    in General Setup

    Posted

    Can you please tell me what is the BHCC and BHCA values currently available in PBXnSIP? Our customer is using a dual core P4 server with 4 GB RAM.

     

    That number fits into the switched telecom world. In IP world it is impossible to present a reliable number, as there are so many factors that influence this:

    • Is the UA using TLS?
    • What kind of challenge mechanism is used?
    • How many codecs are bring offered?
    • Is the mini-session border controller involved?
    • Does the Pc have hardware support for IP?
    • Is VPN being used for Internet traffic?
    • Is there UDP fragmentation?
    • What OS is being used?
    • How big are the routing tables?
    • How much cache memory does the CPU have?
    • And so on.

    There is a page on the Wiki that touches the topic (http://wiki.pbxnsip.com/index.php/Hardware_Requirements), but you will not get a satisfactory answer. An I think looking at the above question, it is really not very serious guaranteeing a realistic number.

     

    BTW there is a setting that limits the BHCA on UDP transport layer. This is just to protect the server against DoS.

     

    It is a little bit like asking "How many pages can you print with Microsoft Word per second" - it also depends... ^_^

  21. CDR Output

    in General Setup

    Posted

    There is already a difference between the timestamp when a call was answered by and agent and the timestamp when the call was connected. The difference is the waiting time in the queue (if the queue connects the call immediately, which is programmable). See http://wiki.pbxnsip.com/index.php/Simple_CDR_Format.

     

    But I must say I am also not 100 % satisfied with the statistics that we get on the ACD. I think the best would be to shoot an email for every call. The email per call has the big advantage that the load gets distributed over the day, and we don't have this huge email at midnight. Especially very busy call centers love to have these statistics, and then the huge email is a problem. Then the midnight email will really just summarize the day and for the details operators need to look into the emails.

  22. CS410 Calling cell phone/ call drops

    in Embedded

    Posted

    I thought I would wait awhile before asking about this, I have a CS 410 set up to call my cell phone when a message is left in my mailbox...the system hunts the phones, then just my extension, then my mailbox, the system then calls me and I answer, if I do not respond to the first prompt imediately, the system will drop the call, so I have to punch in 1 real quick or no call, I dont even get a chance to listen to all the options..if I respond to the call on the first prompt I can listen and do everything I want to...

     

    I thought this might have cleared up with the new software up grades, but this problem continues..anybody else seen this or did I forget to put something into a box somewhere?

     

    The problem is that the dialling on the FXO takes longer to finish than on a SIP trunk. That time is "stolen" from your time to make a choice. Fortunately, there is a global setting "cellphone_timeout" (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File) that defines this timeout. Try setting it to something like 25 or 30.

     

    Also still having an fxo port1 hanging about once every two days and still got the double line thing happening about every 3rd or 4th call...

     

    Did you try to change the polarity change detection and busy tone detection? Those parameters are a pain, but the nature of the PSTN protocol requires to "hint" to the PBX when a call should be regarded as disconnected.

  23. Transfer from Cell Phone

    in General Setup

    Posted

    Yes I do realize that it is on a trunk, but the same trunk is also connected to the pbx at the same time.

     

    I thought that the Inband DTMF detection: On/Off setting had something to do with this feature.

     

    I mean that the PBXNSIP would monitor the call/trunk/extension looking for *77 to initiate a transfer.

     

    Since this isn't the case, what then does the Inband DTMF detection: On/Off actually do?

     

    The PBX does not accept transfers from trunks. This is because otherwise it would be a huge security hole (transfer the call to somewhere in Cuba).

     

    We could make an exception because the PBX knows that the call is to a known cell phone (and it also knows what user to charge). It is just that it is not in there. And also, currently the cell phone would have to put the call on hold first - that is practically not possible with today's cell phone providers. That would have to be changed as well.

  25. Trunks still in Accounts

    in General Setup

    Posted

    I'm running PBXNSIP on Windows version 1.5.6. I just removed an old Trunk (Sip Gateway) that had 5 co lines. ab1 ab2 ab3 ab4 ab5. The new trunk is working fine, however the old co lines still show up under accounts after deleting the trunk. Tried a reboot. Not hurting anything but just taking up license spots. Any ideas?

     

     

    Side question. When you call my main line I now have music playing. It seems to be hold music. The incoming trunk points to hunt group 700, 700 rings for 30 seconds on the main phone and then goes to 701 or 702 depending on time of day. Both of those are auto attendants. Everything is working just fine in terms of calling and rolling. Just plays hold music on incoming calls. If I call from internal to 700 it rings. External I immediately see the phone incoming call light up when music plays.

     

     

    Thanks

    Brian

     

    Yea the references are really done using the literal text "co1" etc. If you remove the source, it breaks the reference.

     

    That with the music is strange. You are sure that the call does not run into an agent group? Or maybe you have a funny SIP device that initiates a call in call hold mode?

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