Vodia PBX
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its has been days but i havent heard anything from your side
is it possible also to fix this problem or should i forget about support
Go to System/PSTN Gateway. There you find settings for Tone 1 (Busy) and Tone 2 (Dialtone). Check if the dial tone settings there match the local behavior in Kuwait. Usually it should be enough to turn Busy tone detection on. After saving, you need to restart the system.
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Any idea? Is there anything we can try on the appliance?
Well, the PBX is just running the standard dhclient3 that comes with the debian distribution. If you can can, try to use a Ethernet hub (or do port mirroring on a managed Ethernet switch) and get a Wireshark trace. Then we can see what is the problem...
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Is a Manual State Change accessing the WEB PAGE as the domain administrator?
Or is a Manual State Change dialing the FLAG and are restrictions placed on which extension can dial the flag?
The way I see it dialing the FLAG resets it, but not means other than WEB or SCHEDULE time to set the flag at this revision level?
I was talking about dialling the flag. You can set the restrictions in the web interface.
Maybe just set up a flag with a 5-minute service duration and try things out...
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We have two phone lines connected to the cs410 and both of them will go to our hunt group. We configure three polycom phones that have line 1 for hunt group 100 and line 2 for hunt group 200. Is there any way to configure the inbound calls from FXO1 going to hunt group 100 and from FXO2 going to hunt group 200? This way we will know that the inbound call is coming from which PSTN lines.
The reason is we would like to know whether the call is coming from our customer or our internal people. Since we always give line 1's phone number (FXO1) for our customer and line 2's phone number (FXO2) for our internal people.
That can be done. The FXO gateway has four DID numbers that will be in the To-Header that identify the line being used. You can use the rule in http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk to route the inbound call then.
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Can you explain how the service flag now supports dynamically being changed?
The PBX just accepts manual state changes also for automatic flags. Then when the automatic transition kicks in, it keeps the state if the desired state is already present.
In other words: If the secretary shows up early in the morning before the flag is actually active, she can dial the flag number and then the day mode is on already. Later when the automatic flag change would fire, the flag stays where it is.
Things are a little bit more complicated in the evening. If she decides to stay later, she has to wait until the flag goes to night mode, then dial the number to change it back to day mode. Before she goes home, she needs to set the flag to night mode manually again.
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What did you put in there? Just one extension?
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after much testing.... I find changing the settings to "Presence Agent" does not help either....
The status is online updated correctly upon startup..... any other time its not working right....
Please advice...
You got any SIP traffic for the update that does not work?
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I am sure they took it out. I even found documentation confirming it on their web site. I have been using the * code for now, I would just like something a little better since the customers love the look, sound quality, and stability of the phones.
I tend to agree. Plus an attended transfer into a conference that has a PIN really makes sense. You can call an external party, then dial into the conference and perform the attended transfer into the conference.
I guess we have to put this on the feature list.
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So provisioning additional settings like dial plans, address books, etc... is done through the spa_phone.xml file correct?
Well, if you have something working now you can start doing your own changes. Now you can take the SPA documentation and drill deeper from there.
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We have install 4 Polycom Soudpoint IP 550 in our company. Periodically, we are seeing 500 internal error message from Polycom Phones and we don't know what is causing that. Here is the logfile we are seeing the error message.
The "SIP/2.0 500 Internal Server Error" is for the NOTIFY and usually it has to do with subscriptions that are still in the server but already expired on the phone. After a reboot this is normal if the phone comes up within the subscription duration and there is no reason for concern.
The reason why the call gets rejected is "SIP/2.0 488 Not Acceptable Here". In the INVITE you can see that the PBX obviously does not offer any codec to the phone, which makes that understandable. Check your codec settings for the system, it seems the problem is there.
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So far there was no need to have attended transfer into a mailbox. Are you sure those phones don't have blind transfer any more (check the more button)? I can't imagine Cisco took such a basic feature out.
Workaround would be to use the *77 transfer code, but that would make the phones really look extremly basic.
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Try http://192.168.32.30/provisioning/spa_phone_$MA.cfg. Then the HTTP server knows that this file is for PnP.
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Is this feature in the 3.0 CS410 releases currently available?
I checked the WIKI and see no references.
The builds for CS410 that are younger than one month have it. Documentation is still on 2.1.
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If I just give the phone the IP of the pbx:
Profile Rule: 192.168.32.30
Then the phone doesn't register. In fact there are no log entries so it looks like it doesn't even try.
Hmm. Maybe better forget about TFTP... Check out the attached files (you can also put them into the html directory, if it does not exist yet create it). There it says you should out http://192.168.1.2//spa$MA.cfg (if 192.168.1.2 is a PBX IP address) into the Profile Rule.
I assume you have put a MAC address or just a star into one of your extensions? There is some general information at http://wiki.pbxnsip.com/index.php/Prepare_...r_Plug_and_Play.
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I am trying to setup call accounting, and I am having a problem seeing the extension of the conference bridge if a call comes in, goes through an auto attendant, and then into the conference bridge.
My CDR format is:
$m $e $b $B $d $o $c $f $t
The output I am getting is:
voip.office.twincitytelephone.com 20080604 163107 42 <sip:4193922384@voip.office.twincitytelephone.com> conference
I would expect to see the conference bridge extension number right between the domain, and the date.
Maybe the workaround is to stick to the "conference" string for clear identification it was a conference call.
In 2.1.11 we'll change the type to 'v' because 't' was supposed to give the To-header. The extension is empty because the call does not go to a registered extension.
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Doesn't look like the SPA942 has tftp options via the interface. Is it just a matter of specifying the tftp port after the IP of the pbx?
Just put the IP address in... Then the phone will automatically choose TFTP.
And by setting passwords - do you mean the extension passwords? There is no place to specify password in Admin/Ports/TFTPThe setting is called "Allow TFTP Password" - just set it to "always", then later when everything works you can lock this down.
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Does anyone have the lines necessary to add to the provisioning script to auto provision the Linksys SPA942 phones?
Just point the tftp server to the PBX. It should work already... Also make sure that you do provision the passwords (Admin/Ports/TFTP). The problem is here that Linksys requires their own little secret algorithm for encrypting config files, which they won't gibt to use (the secret, not the program
). Also, it helps if you are using Option 66 on DHCP.
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Check out http://www.dslreports.com/forum/r20408090-...wn-Since-Sunday, maybe the free service is just not available...
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After doing some research here, i found that the problem is with CO lines. If i remove all the CO lines from our system, the incoming calls roll out to the cell as expected. If i put the CO lines back, the calls immediately hang up.
I'd really like CO lines so we can monitor what lines are in use on our SNOM360's. But in the short term, removing them has my system working.
So you are sure that there are enough CO-lines on the trunk? Check the account overview on the web interface, if the PBX seizes a line it will show it there (you'll see the caller-ID associated with the CO-line).
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btw. the number 59 under RdrctNum=59 is the one we use in the trunk for the OCS Mediation Server. Setting "Assume that all calls come from user:"
The "Assume that all calls come from user" was a workaround to get things working at all - the Caller-ID presentation had lower priority. But maybe there is a way to configure the PSTN gateway to use the From-header for rendering the Caller-ID.
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However, I would like to know How to get the correct "presense information" displayed on the other party?
I do not see the icon change in xlite from something offline to online... regardless of whether the other party is registered or not....
For that you need to put the party into the contacts list. There you can also see your own presence status. We tried it, and it works okay. The PBX is just a relay of the status, not looking into it.
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... the only thng now is that the caller can press 1, 2 etc while hearing the ringing sound.
Well, for time being call that a feature
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We have a 5 static ip plan (4 available). However, the WRT54G obviously isn't capable of muli-wan ip's.
Is there an affordable and reliable router (with at least some basic QOS) you could suggest?
If you are just using IP on the WAN side, it might be an option to use the WAN interface of the CS410 for the public IP. Then you just need a hub/switch to connect the WAN to the router and the CS410. Then the CS410 will use one public IP address (and a private IP address, make sure it is a static one to avoid problems with the default IP gateway) and the router will use another public IP address. However in this setup, QoS will remain a problem.
In the ultra-low cost segment, I only know about OpenWRT project where you essentially load Linux on the router. That gives you a lot of options if you are able to setup Linux routing.
We did search for low-cost router solution some time ago. In the end we gave up on it and bought a standard Cisco router on eBay.
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I can't find (in PBX, Wiki and forum) how to set a delay before the AT picks up, now it directly answers but it causes things like somebody doesn't here the complete message.
So I want to hear the " ringing" tone for e.g. 5 sec before the AT picks up, where can I set this option?
AT the moment there is no such settings. IMHO it would not be a auto attendant setting, as the same problem also exists with all other types (e.g. mailbox, IVR node). And it also depends on who is calling, e.g. a call from an extension to the auto attendant might be very very fast while a call from a trunk might be very slow. Maybe it would be a trunk setting. OR maybe it even has to be a setting of the PSTN gateway?
).
Statiscally register an external extension
in Extension Setup
Posted
That should have been fixed in 2.1.3 (see http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.3). What version are you on?