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Vodia PBX

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Posts posted by Vodia PBX

  2. Help needed

    in Fax Setup

    Posted

    Hmmm, I don't see an updates from anywhere at all. I assume I should see this at the point the sipura detects the fax and should send an UPDATE to the PBX to change negotiate T.38?

     

    No, it's just that the UPDATE implementation of most devices is actually broken and we had the same effect as you - until we turned the support of UPDATE off.

     

    And yes, if the FXS detects FAX it should send a Re-INVITE with T.38 in it. If it does not then that is a sign that the T.38 is not enabled on the device.

  3. No audio at all?

    in General Setup

    Posted

    I have CS410 at two locations . . . setup everything and I can call one location and phone rings, answers, but no audio either way. I took a phone home and tried it there, I can get "we're sorry" message from telco on the phone, but no audio otherwise. I'm pulling my hair out here . . . neither my simple WRTG54 nor my larger 16 port router are working right . . but I was able to get my asterisk box through the system before. Is there some troubleshooting things to look at -- and yes, I've looked at the WIKI on one way audio and that wasn't very useful at all. The phones are SNOM 360's on the latest firmware 7.1.30.

     

    How is your IP config? Are you mixing DHCP with static IP addresses? Maybe it makes sense to log in through SSH and take a look at the IP config with ping.

  4. Smon360 address book & language issue

    in snom Phones

    Posted

    I have reset the phone to factory defaults and reprovisoined them, I have even tried to install the pbx on a new clean machine but the phone always get the first 9 entry of the address book but not the remaining.

     

    Did you try to enter a key? Maybe you are in the interactive mode where you have to enter a key to drill deeper into the address book (e.g. 2 means ABC).

     

    About the language issue, the lang files are in place and the phones load them, but I need to select and use the Italian when the phone gets provisoned and not have the phone proviosned with other langages but still defaulting to english.

     

    We need to take a look at this here. Maybe the problem is with Italian language, not so much mainsteam B) on a global perspective...

  5. Passing Parameters to Snom Phones

    in Other best practices

    Posted

    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, in those cases you should look into the generated directory and pull out the files that you want to modify and move them into the tftp directory. Then you can edit them there and add or change parameters.

  7. Several Questions on Conference

    in Conference Room Setup

    Posted

    Regarding star codes *9 and *1, the funny thing I found is that every participant can dial *9 or *1 except moderator. That is, every participant entering conference room by dialing participant's PIN can dial *9 and *1. But every participant entering conference room by dialing moderator's PIN cannot do that. That happens on all 2.1.x versions including the latest version 2.1.10.2474.

     

    Ops, very good catch! The PINs were mixed up in the web interface. Emails seem to be okay.

     

    Regarding the e-mail on conference # and PIN, I did try to put both participant's extensions and e-mail addresses separated by ';' into the Participants field and no e-mail was sent out by the system.

    The other thing I found is that the Start time and End time of scheduled conference don't work as they are supposed to. I can enter the conference room before Start time or after End time. I think this might be a bug.

     

    Well, the interface is very lax on the start and stop times. Essentially the whole conference scheduling is about agreeing on access codes and being able to send an email out that Outlook can read. There is no resource reservation, and if people want to start earlier or later, that's fine with the PBX.

     

    I think that is a feature.

     

    Anyway, seems like we'll need a 2.1.11... There is also some stuff for E911 that needs to be addressed.

  8. Caller-ID Issue

    in Embedded

    Posted

    We are the 4th unit in a row of buildings and the junction to the phone company is in the first unit. There is probably about 100 feet from the appliance to junction in our unit then another 200-300 feet to a second junction box and finally another 100-200 feet of cable to the junction where it connects to the phone company's cable. So, I wouldn't be surprised if we have some poor wiring. We had issues getting our T1 line set up and we have had problems with our DSL line and some coss-talking with an unknown line (not one of ours) in the past.

     

    Hmm. Before changing the cable, maybe it is worth to play with the input gain and/or get a FXO signal booster. Maybe there is also a simple way of getting the cables measured out, the good news about FXO is that it is stoneage technology and there is so much stuff out there for this.

     

    If it is a cable related issue then it will take some time to fix. I can replace the cable inside our building but I will have to ask the landlord about the rest of it. In the mean time, is there a way for a caller to get Direct Inward Access like that provided when it recognizes your cell phone number?

     

    Well, those features rely on the Caller-ID detection... The only thing that you can do is setting up a "calling card" account and dial into it (maybe through the AA), then enter the extension number, PIN and then you can place outbound calls from there.

  9. Caller-ID Issue

    in Embedded

    Posted

    Using the PSTN FXO lines it seems that our CS410 only picks up the Caller ID sometimes. I have been testing it with my cell phone (trying to get the special autoattendant to pick up). I had this issue with 2913 and now I just updated to 2914. It seems that after a reboot it will pick up the Caller ID at least once.

     

    How long is the line? What is the carrier? "Sometimes" sounds like a analog problem. Also, set the log level high and turn PSTN logging on (after that you need another reboot). Then you may see the messages about the Caller-ID in the log.

  12. CO Line Pickup

    in Extension Setup

    Posted

    On snom phones if you have a button monitoring the CO lines, you can put a call on hold, and the CO line will blink on the other snomes. If you go to another snom phone, you can pick up the call by pressing the CO line that the call is holding on. My question is if you want to pickup the call holding on the CO line from a non snom phone, or one that does not have buttons associated with the CO line, can you do it? I have my CO lines numbered 701 and up. I cannot dial *87701 to pickup the call from another phone.

     

    Do you want to pickup or retrieve the call? Retrieve is *86701...

     

    Star codes are not very popular because they are so hard to use!

  13. Several Questions on Conference

    in Conference Room Setup

    Posted

    1. What's the difference between moderator and participant? I notice there are different PIN's for moderator and participant. And according to the your wiki website, the moderator has the ability to clean all participants of the conference by dialing *9 during the conference. But what I found is that every participant can do that as well. Is this a bug? I am using version 2.1.6.2447 on SuSE 10. Maybe I should upgrade the software.

     

    There differences are few in the conference. The moderator has the right to kick everyone out (*9) and send a email to himself who's in the conference (*1).

     

    2. How can I configure the system to send e-mail including conference # and PIN to each participant? I put each participant extension into the Participants field when creating scheduled conference but the system didn't send e-mail to each participant extension.

     

    You need to seperate that list with semicolons and of course the email settings must be okay for the domain.

     

    3. Where can I find the latest document on pbxnsip? The document available on your website is too old (version 2.0.x).

     

    http://wiki.pbxnsip.com/index.php/Conferencing and http://wiki.pbxnsip.com/index.php/Scheduling_Conferences are the latest and greatest.

  15. CS410 Issues

    in Embedded

    Posted

    Why do the lines show hot via the LED's but not on the web interface?

     

    Well, the "CO lines" on the web are far away from the real hardware. If the lines are still on, the FXO gateway obviously believes that the call is still on. Anything in the log with PSTN? You can turn logging for PSTN events on (requires a reboot, unfortunately).

  16. CS410 Issues

    in Embedded

    Posted

    Calls dropping are usually a problem with the hangup detection. Unfortunately, FXO is not very clear about when to hang up a call. In an extreme case, an operator might play the message "The other side has disconnected the call, now it is time for you to hang up" and wait until the PBX disconnects the line.

     

    There are three ways in the CS410 to detect call disconnect:

     

    Detect Busy Tone: The PBX tries to detect a busy tone on the line. If the other party plays a busy tone (or something that is similar), well then the PBX things oh that's my time to disconect the call. If your operator does not play busy tone, then shut this off.

     

    Detect Dial Tone: Similar, but with dial tone. In doubt, turn this off.

     

    Detect Polarity Change: That's another way. The idea is to change the polarity of the analog signal when the call gets connected and when the call gets disconnected. This can be dangerous if you have a long line and the detection is on the edge.

     

    If you know how the operator indicates the call disconnect, then you should use only that method.

     

    BTW the latest and greatest is http://www.pbxnsip.com/cs410/[color=&qu...update-2914.tgz[/color]

  17. One trunk with several numbers

    in Trunk Setup

    Posted

    I think I understand the patterns but lost on where this gets placed. Would it go on the trunks page in the extension field? If so does it sort in order and the use a back up at the end? 300 would be the auto attendant.

     

    Example exact typing in field being

     

    Extension: !1([8128675308]*)!\1!t!301 !1([8128675309]*)!\1!t!302 300

     

    Those brackets are probably not right. Maybe you wanted:

     

    Extension: !1(8128675308)!\1!t!301 !1(8128675309)!\1!t!302 300

  18. Time Problem

    in Embedded

    Posted

    That took care of the issue. It is now pulling times correctly and setting service flags at the right times. Thank you.

     

    However, now I have a different issue (though this might not be the right place to post it). When an extension picks up an incoming call, the auto attendant plays during the conversation. My secretary told me that on an incoming call today it started playing a few moments after she answered the phone. I have had it happen to me as well. I answered the phone, the caller was there and it played the auto attendant message. We have a recorded message for the autoattendant

     

    I'm thinking that the caller is hitting more keys then they need to but it is difficult to tell. I asked my secratary to let me know if it happens again immediately so I could save the log from the call.

     

    The cross-talking could be a problem with the analog lines. Maybe there is a line connected twice.

     

    Also make sure that you have the RTP ports for the PSTN gateway in a good range (e.g. 2048- 4096).

  19. Smon360 address book & language issue

    in snom Phones

    Posted

    Hello,

    I have two problem with pbxnsip 2.1.10.2474 e snom 360 fw 7.1.30, I can autoprovision the phones but when I press the directory button on the phone I get only the fist 9 entries of the domain address book (about 20 entries) and the following message in the logs:

     

    2008/05/23 12:18:59: Web Server: File snom/adrbook.xml?user=1&auth=557fa927fedc754de431e50f3688e89e not found

    2008/05/23 12:18:59: Remote site closed the connection

     

    Second problem is with the language, I can provision the phones with the available languanges but I dont know how to set it to Italian on autoproviosion I have to manually select it.

     

     

    Thanks.

    valerio

     

    Regaring the address book: Did you change the user, the domain or the password? If yes, you need to re-provision the phone.

     

    The second language requires the appropriate lang_xx.xml files from snom in the tftp directory.

  20. Help needed

    in Fax Setup

    Posted

    Do you see anything with UPDATE? That was a common problem with some devices that indicate they support it, but when you send UPDATE nothing happens.

     

    Otherwise, Wireshark is most useful in these cases... Maybe you can send my a PM and we'll take a look at this.

  23. Time Problem

    in Embedded

    Posted

    I just purchased the CS410. I updated it to the firmware to the latest version aviable at the time. Currently running 3.0.0.2913.

     

    The time on the appliance is always incorrect. I have tried logging in and pinging the NTP server and I don't get a response. (The DNS resolves because my Small Business Service is also our DNS server but no response back).

     

    If you are using both Ethernet interfaces make sure that you have only one IP gateway set. This is a problem if you are using DHCP on eth0 and static IP on eth2. In that case you can solve the problem by using statc IP on both ports or by telling the DHCP server not to provide the default IP gateway for the MAC address of eth0.

     

    But you are right, PING must work, otherwise it is hopeless.

  24. One trunk with several numbers

    in Trunk Setup

    Posted

    What is the best method to setup several lines on one trunk.

     

    Scenario being Trunk has 3 phone numbers with on registration. I only see one place to route the trunk to. Is there an alias or addition you add to te trunk settings to specify where each number goes?

     

    One lines goes to ext 301 which is an auto attendant

    Second line goes to 302 for a fax

    Third line goes direct to an Exec. I read through the incoming routing on the Wiki but I'm not following.

     

    Maybe take a look at http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk. Depending on your service provider, you probably find the destination in the To-header. In your case, maybe a pattern like "!([0-9]*)!\1!t!301" make sense.

  25. Moderator Adding Participants

    in Conference Room Setup

    Posted

    Is it possible for the moderator to add a participant to a conference call, either in Ad-hoc or in Scheduled mode?

     

    It is possible to blind transfer participants into the conference. At the moment the ugly part is that the participants have to enter the PIN code for the conference after they are transferred.

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