YMSL
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Posts posted by YMSL
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Is their a list of all the directories created during fresh installation and what they are used for?
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We have a windows version (not GA yet) that takes care of the recording related issues. You can download this from http://pbxnsip.com/protect/pbxctrl-3.3.2.3183.exe
All calls get recorded but incomin and outgoing have no differences on this built it always put -i- does not matter if is incoming or outgoing!
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I have attached a wav file to let you hear the sound of Joe's voice..
It sounds a little like Darth Vader.
Does anyone know what this sound is an indication of?
Once he did a power reset on his phone (Aastra 57i) the trouble went away.
I myself have also experienced this and had to reset my phone too.
Any ideas?????
mmmmmmm, we got the exact same thing, exacte, we've found taht a device was installed on the network (online faxe service) on a local dedicated machine. When it was turned off the problem stopted.
It might have been a pure coincidence but since we move the PBX installation to a dedicated Internet service we never had it back.
do a sharktrace if you can, you might find very interesting things...
hope this helps
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I have pbxnsip 3.3.1.377 (Win32). On Domain -> Accounts -> Redirection, I have the following:
Cell phone number: <my cellular number>
Confirmation: No confirmation required
Include the cell phone in calls to extension: Immediately
Specify time when system calls the cell phone: No specific time excluded
When I get incoming calls, both my Polycom SoundPoint IP 670 and my cellphone ring. The cellphone even shows the correct caller ID. However, if I answer the call on my cellphone, I get nothing. The call appears to be connected but there's no audio in either direction.
I can make and receive calls on my SoundPoint to the outside world so there doesn't appear to be a problem with my Broadvox trunk.
Any suggestions as to what might be causing this? I'm answering the cell on the first ring.
Thanks in advance.
PS: this has been a problem for me going back a while.
We got the same thing and it was a codex issus between gateway and PBX
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We had a similar complaint sometime back and an instance of PAC running against the PBX was causing the issue. After they closed and restarted the PAC, things were fine.
Same here yesturday around 2PM on the 28th.
reboot, killing the app, removing the licence and try the 3 minutes... nothing, aboslutelly nothing worked...
I have manually disabled pbxctrl and manually uninstall .NET Framwork 3.5 upgrade and 3.0 upgrade, reboot the machine and restart the server, everything seams to be fine since...
I suspect that something is going wrong with pbxctrl since it is targetting a specific processor and when stopping pbxctrl targeted processor is getting back to normal
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I believe VNC can do it; however did not have the chance to try it out.
The CS410 uses this mode internally to receive the media from the MoH plug.
NOT working. at all!
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Hi All -
I am looking at rolling PBXnSIP at remote location which already has a few POTS lines in place. While I had considered using the CS410, I would prefer to use our own server equipment as our team is trained on them, they have backup processors, power, etc - (...and they are already installed and in place!)
Has anyone had good luck with a specific brand of VOIP gateway or is one kind recommended for use with PBXnSIP? I found a long list at:
http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways
...but it's a long list and I have no frame of reference.
Again, I only have 5 or 6 POTS lines and just want to make them usable by PBXnSIP. I imagine this box would simply make these POTS trunks available as SIP trunks.
Suggestions anyone?
Thanks,
Alex
1- SPA3102 Cisco/LinkSys is not working very well
2- Mediatrix have a VERY VERY VERY VERY BAD SUPPORT but the device is kind of working fine! BUT VERY BAD
3- AudioCodes is triky to set up but work nicely
4- Vegastream is expensive for waht is does but run smootelly!
here is my little experience
:-) Good lock
I go with VOIP provider over a T1 Internet connection, worke fantastic but a bit expensive!
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You can put a one-person hunt group in from of the AA (the only person in that hunt group would be the secretary). However you will obviously loose the FAX functionality, unless the secretary wants to detect the tones and redirect the call to the FAX...
Got it thanks. That is why I was truing to use an IVR to detect fax and voice and to root accordingelly!
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Oh the AA does not use regular expressions. Just put a "F" into a "User Input" and then the FAX extension into the "Destination". The AA will detect the FAX tone similar to a DTMF tone with the name "F" and will behave as if the user entered "F" during the annoucement.
Many thanks for your reply,
It is working and It was easy!
What should I do it if during business houres we have a humain receptioniste and all incoming calls goes to here? IVR? How?
many thanks
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Where do I input this !T!1104! !F!1111! on AA I cannot find any logical localisation for that setting ???????
anyone!
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Many thanks for you input,
I am sorry I do not get it!
so on my trunk putting
Send call to extension: !T!1000! !F!1111!
where 1000 is an AA
and 1111 is the direct extension for FAX ?
I have tried
Send call to extension: !T!1104! !F!1111!
to test it, where 1104 is an extension and no matter what Mediatrix is sending it to 1000 (AA)
but If I set it up to 1104 (only) it work!
many thanks
Where do I input this !T!1104! !F!1111! on AA I cannot find any logical localisation for that setting ???????
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So does it already go to the IVR node?
Try using the auto attendant instead of the IVR node. Depending on your DTMF setup, the IVR node might not be able to detect the CNG tone. As you are going to the AA anyway, that should be no big change. Then you can use the "F" in the direct destinations.
Many thanks for you input,
I am sorry I do not get it!
so on my trunk putting
Send call to extension: !T!1000! !F!1111!
where 1000 is an AA
and 1111 is the direct extension for FAX ?
I have tried
Send call to extension: !T!1104! !F!1111!
to test it, where 1104 is an extension and no matter what Mediatrix is sending it to 1000 (AA)
but If I set it up to 1104 (only) it work!
many thanks
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Good day,
I have 4 PSTN line on a mediatrix 1204
number 1: 4185501234
number 2: 4185506789
number 3: 4185506781
number 4: 4185506782
number 5: 877xxxxxxx taht root call to a cascade on VOIP DID
and 1 VOIP DID (with capacity of 50 concurence) used for Voice only
I am usign Number 2 (4185506789) for fax
I want to set PBX to detect if it is a fax or a voice call and root accordingelly
So I have set up my Trunk with the following:
Send call to extension: !4185506789!1001! 1000
where ext 1001 is an IVR with :
DTMF Match List: !T!1000! !F!1111!
and where 1000 is an AA
so fare it is not working, what Am I missing
V. 3.1.2.3120, cannot use latest due to problem with all call recording function of CallCenter edition
many thanks
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Deleted that user
many thanks for your quick response
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this user is posting website and information to promote is XXX site. Please advise
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hi,
I'm pretty sure the push button call record works in 3.2 because i use it.
Is this record all calls feature?
tx
matt
We do not use push button to record, this costumer is usign Cisco phones
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Good day,
Since v 3.2.0 we had recording problems with CallCenter version (win32).
Version installed 3.1.2.3120 - system record all calls like requested
version 3.3.1.3177 does NOT record calls if call is coming from PSTN line and terminate to internal extension.
This problem was reported on V 3.2.0 and still not working on 3.3.1
moving from 1 version to the other using upgrade - no settings changed
Please help
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Since you have 9:00-18:00 on the service flag, this account gets activated during that time.
Either you can modify the service flag account to be manual and then set the state to 'clear' Or you can change the time range to something outside your normal office hours. This way, auto attendant will behave the way you wanted.
This is conflicting, form my point of view,
service flags should have settings something like
set to open hours or set close hours
On 1 costumer case we have to set service flag to 0:00-8:59 12:00-13:00 17:01-23:59 in order to have humain receptionist during business hours and AA when close.
jsute a remark
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Any update?
Is it fixed on 3.3.1? download link?
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No. Let us check what is the problem.
I have set up a test system with a demo key and I have the exact same problem (win32) and audio recording compressed
So where do we stand?
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I would like to know if it is possible and how we can set up call forward to dial an external number that include a pause in the number.
something like a cell phone
1 555 555-1212T101
many thanks
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Running Win 32 3.3.0.3165 Callcenter 50 License duration: Permanent
I do not know what I am missing but the system do not record incoming calls if it is comming from AT then transfert to an extension.
my settings:
Domaine set to:
Recording default for this domain: All set to Yes
All extensions set to:
Recording: : all set to default
Any ideas on what can be wrong, am I missing something?
I went back to V 3.1.2.3120 and everything is back to normal?
Do I need an new licence?
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Just like YMSL, it is not completely clear to me what you want to achieve.
I would just like to inform you that it is possible to use the SPA3102 firmware 5.1.5(GWa) as a fully functional Gateway and SIP-FXS-client at the same time
If anybody is interested i could provide you with the configuration...
Does anyone have access to this particular version SPA3102 firmware 5.1.5(GWa).
With 5.1.7 or 5.1.10 FXO never register (FAILD)
The only version I can have fully working is 3.2.6(GWa)
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HI,
I need to set a new PBXnSIP small office system and looking for proper part before. What gateway would be suitable for 2 FXO only? For 4FXO I would be usign Mediatrix 1204 but for 2 only what would be good and not expensive?
many thanks.
Does SPA400 is know from PBXnSIP?
Signal - max cable length
in Linksys Phones
Posted
Is they're a limit to the length of cable in the signal quality?
grade 5 cabeling with Cisco POE switch.
If yes, what should be done if cable are 500 feets longs?