Vodia PBX
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Do you have an updated version of the template for 3.3.0.3165?
That's the latest and greatest.
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Ok, then why when I called after hearing nothing and calling from my cell did it give me a busy signal??? The call was probably terminated because I or the client couldn't hear anything. Never rings or anything. When calling from cell it also never rings but we get busy signal audio.
Nothing = digital silence? Or did you hear "comfort noise"? Comfort noise means that the carrier is trying to locate the subscriber - which can take a while in the cell phone world.
The PBX also plays comfort noise when it tries to locate a SIP user. This is usally pretty fast, but it can also take longer and then the PBX also sends comfort noise to indicate the caller that the line is not dead.
Some people feel uncomfortable with comfort noise! They are confused that a digital modern system can have a noisy line... This is really not a bug, it is a feature.
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I have had some compliants that calls aren't completing but when I check the calls are receiving 487 Request Terminated which I believe is a busy signal because when I call that is what I get from my cell. This is something that has been haunting me for weeks.
Please let me know how we can provide busy signal audio for this.
487 is not busy, it means that the request has been terminated - usually by the caller after the caller sent a cancel.
486 means busy.
A SIP trace would be useful in this case to see what is going on.
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What does the line:
[5] 2009/03/23 19:23:24: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT
indicate is happening?
Well, that says that the PBX received a caller-ID (either through DTMF or FSK).
Hmm. Do you mean that the disconnect is actually okay, but the detection of the caller-ID is the problem? Maybe the carrier wants to send some information after the call was disconnected?
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The pots cable from the modem is about 1 foot long.
I think we can rule out gain and noise problems...
Can you log into the box and try the following?
cd /etc/init.d
vi pbxnsip
then edit the line that starts the PSTN gateway:
$SIPFXO --config /etc/sipfxo.conf --show-version /etc/sipfxo-rel
ease --pbx-adr 127.0.0.1 --sip-adr 127.0.0.1 --csp-mac "$MAC0" --msp-mac "$MAC
1" >/dev/null 2>/dev/null &
Add the option "--ignore-no-loop" to the options. That should disable the loop interrupt detection. Lets see if that shuts the loop interrups down.
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I have a customer who is having issues adding in a 3rd trunk to his CS410.
I have run the license through a decoder which clearly states "Trunks = 3".
There is only 2 trunks configured! 1 trunk to an ITSP and 1 to a SIP g/w.
Upon trying to add a 3rd trunk the system gives an error stating that trunk limit exceeded.
Currently running f/w v.3.3.0.3165, the same error occured on f/w v.3.1.2.3120 & 3.2.0.3143.
In my experience with CS410's i have seen inconsistencies with this which can allow more than 3 trunks to be configured without any additional trunk licenses.
Anyone else come across this issue?
Does the CS410 count the PSTN g/w as a trunk even if it is deleted from the trunks list?
It could be as simple as the license code does not allow more than two trunks (it is base64-encoded, so it is easy to see what it says about trunks).
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Alert-Info: <http://127.0.0.1/Bellcore-dr2>
That means that the PBX was not able to detect the Polycom phone during the registration. Does the phone sit behind something that changes the User-Agent string?
From ringtones.xml:
<vendor ua="Polycom.*" type="alert-info">Auto Answer</vendor>
Seems that the user agent seem in the REGISTER message does not start with the string "Polycom".
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I keep getting an error from my sftp program when I try to manually copy the tftp files over. I even killed the pbx to see it was doing it and it still wouldn't let me drag and drop the files over and I have pleny of space after deleting the old ones. What would the procudure be to create an update.tgz file that I could pack the latest polycom firmware. That would be nice to see on the download page!
The first step in that procedure would be adding more RAM to the box. Unfortunately, Polycom firmware is in the 50-100 MB range.
Welcome to the embedded world.
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Does SPA400 is know from PBXnSIP?
I believe that the SPA devices use more or less the same software. That means it should work fine. Plus the feature set from a PSTN gateway is pretty limited anyway. And the investment is also limited - so IMHO it is worth a try.
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I can certenelly do that, what screen do you need?
Yes, conference room is a scheduled room
The screen of setting up the conference (user mode).
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The CS410 is connected to a Time Warner SBV5322 that provides 3 "pots" lines connected to FXO1 – FXO3. (the 4th line runs directly to a fax.)
CS410 Version 3.3.0.3160
X-Lite soft phones 3.0 build 47546.
The system worked perfectly for 6 months.
That of course raises the question: Has anything changed in the setup?
1) For the last week we have been "receiving" calls from Anonymous (anonymous@localhost) [PSTN] after any call has been ended (from either side).a. None of the FXOx port indicators are lit, but the system does list the "call" on the "Currently Active Calls" page.
b. The line is silent when answered and when unanswered the Ghost calls connect to VM boxes and fill them with blank messages.
c. An email with a RE line similar to:"CS410: RTP Timeout on 81304908@fxo#d1caa153cb" is sent stating that "The call between sip:7608328620@localhost;user=phone and sip:anonymous@localhost;user=phone has been disconnected because of media timeout (120 seconds), 0/5994 packets have been received/sent"
2) For the last month FXOx ports will randomly "lock up" making that line unavailable.
a. The FXOx port indicator will light, but the system does not indicate a call in the "Currently Active Calls" page.
b. If two internal users are on lines, when the third user dials out the system will drop one of the existing calls.
c. Sometimes unplugging the line from the offending FXO port will clear the problem, other times the CS410 will have to be restarted.
It sounds a little bit like the PBX has problems with the FXO. Maybe it was on the edge already and now it is over the edge. Is the signal quality okay on the FXO (I don't know the Time Warner - is it connected through a short cable?).
Maybe you can attache a screenshot of the settings for the FXO gateway on the CS410. It could be there is a parameter wrong.
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CS-410 3.2.0.3143 (Linux)
When and incomming call is transfered to a cell phone the caller hears an announcement "This number could not be found" after a minute or so.
Followed by "Please enter the destination number followed by the pound key".
The Xfer procedure is:
Answer incomming call
Press Xfer
Dial cellphone number
Speak to person on cellphone
Hang up
The caller is then transfered OK and they cal talk to the cellphone until the announcement comes on and the cellphone is disconnected.
Is there a different procedure for this type of transfer???
No, the prodecure is okay.
That sounds like trouble with the connectivity of the call. Probably the PBX believes that the call is not connected yet (you can actually talk also on unconnected calls). Do you see a 200 Ok coming in from the cell phone call?
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Running 3.3.0.3165 CallCenter edition
Loging in as user and going to the conference tab to create a conference. Entering all the info and picking an availible room.
It has absolutelly no effect waht so ever. Email is not sent out, conference is not registering, I see no trace what so ever just like if nothing ever happend
Just tried that... Works here. Can you attach a screen shot? Is the conference room a "scheduled" room?
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Please address the HTTP provisiong issue stated here and above in this post.
Can you start a new thread...? Things get messy in this thread... 3.3.1 is already in the pipeline.
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I am not sure exactly what you mean by making sure all user agent subscribe to MWI event. This is an on-going question that all my users have seen, as well as myself. Is it a known problem? if so is there a resolution?
Are you doing plug and play on all phones? WHat version of the PBX? What phone model? What version on the phone?
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We have a hosted environment and 1 company's time is totally off and all phones don't have the same time. No phone shows a date either. All other companies work fine and even another branch on this company is working as normal. I can have them ping both pool.ntp.org and north-america.pool.ntp.org and get replies with no errors. Has anyone else seen anything like this? Their times and dates where working weeks ago and nothing on our end has really changed. Thanks for any input.
Does their DHCP server provison a NTP server? Many phones will take that server over the privisioned server. If you can look into the log of the phone maybe you can see what NTP server the phone uses.
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I have attempted to record using *98500 (where 500 is my AA). As-soon-as I hit *98 it wants me to record my voicemail greating. I have tried uploading a file using the IVR but that hasn't worked either.
Oh what phone are you using? If you are using Linksys, make sure that you change the dial plan on the phone itself - it starts dialling immediately after two digits if you use the factory-setup.
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I am attempting to use custom prompts with Pbxnsip but am having major difficulty. I have read through the WIKI, documentation and the forums, but I can't seem to find a step-by-step method to make it work. I have uploaded the correctly formatted wav files but I see no way to get the system to use them instead of the default prompts. It continually talks about the service flags in the wiki, but that makes no sense to me.
Can you hear the WAV file? If not, you probably have the wrong format.
Or do you have problems on how to set the AA up? Check the IVR tab. There you can control what other built-in prompts the PBX will play back. These other prompts cannot be changed (unless you decide to overwrite files in the audio_xx directory, which is usually not a good idea).
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When do you expect this new release to be ready?
Beginning of next week.
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This is starting to get a lot of folks very angry. It is tough to troubleshoot because it seems to be random but once it starts on a base the only thing that will fix it is a restart of the base.
Last resort is to get some stuff that restarts the devices at night.
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That would be very nice if you want to test that for me.
Looks like there was a gremlin in the provisioning of the language... I guess we have to put this into a 3.3.1.
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I was wondering if its possible to park a call by doing the following actions:
when i have a call i will transfer it *85, the PBX will then announce me which "parking spot" it assigned it to, for example "your call has been parked on orbit 301" then i will announce on overhead paging, john you have a call on orbit 301, all john will have to do from any phone in the building dial *86 and it will ask him please enter the parking orbit you would like to pick up
if its do-able, i dont see how to do it, maybe someoen can help me
it seems as right now all im able to do is put a call on orbit of the extension, which is ok, but then when i go to retrieve the call, it gives the the "first come first serve"
i want only 1 parking per orbit, and the PBX to tell me which Orbit its parked on
He should pick it up with *86301 if he wants it only from orbit 301.
Or put "*" into the setting "Explicitly specify park orbit preference". Then the PBX will prompt for the orbit on those extensions where you put the "*". So maybe for the one who usually parks the calls put the orbits there, and all others who usually pick calls up put a "*" there.
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Upgraded to 3.3.0.3163 of the PBX, removed my custom polycom XML templates from /html so it uses the system ones. Now the phone time is wrong as the DST start date is MIA ..
tcpIpApp.sntp.daylightSavings.start.date=""
this should have a value of "8" (second instance in the month)
Please fix this so the variable to get that populated works.
** FROM POLYCOM ADMIN GUIDE **
tcpIpApp.sntp.daylightSavings.start.date 1-31 8 If fixedDayEnable is set to
1, use as day of the month
to start DST.
If fixedDayEnable is set to
0, us the mapping: 1 = the
first occurrence of a given
day-of-the-week in a month,
8 = the second occurrence
of a given day-of-the-week
in a month, 15 = the third
occurrence of a given
day-of-the-week in a month,
22 = the fourth occurrence
of a given day-of-the-week
in a month
It should be "dst_start_week_of_month" that is set to "2"... ?!
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Can I still put extra xml files for additional snom settings in the html directory?
And to get the dutch language on my snom phones, do I still have to place these extra files in the html directory?
Yes, you can still put extra files there, but now the name must be snom_320_custom.xml (replace 320 with the model).
I guess I should just try to provision a phone in Dutch and see what happens...
Sending Extension Number on Internal Trunk
in Trunk Setup
Posted
Maybe that trick can help: In the ANI field, you can tell the PBX to use a different ANI when using a specific trunk.
For example, "9787462777 Trunk1:123" would mean: On trunk "Trunk1" use "123", otherwise use "9787462777" as ANI.