YMSL
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Posts posted by YMSL
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I am trying to use Incoming anonymous calls behaviour "Pretend to be busy" to send Incoming anonymous calls to voice mail. When set, it act exactly like "reject call" and tle the anonymous caller that I do not wish to get calls from anonymous callers.
Is their anything I am missing?
PBXnSIP: 4.3.0.5021 (Win32)
It does the same for previous version...
thanks for your help
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Many thanks for your reply,
PBXnSIP, now SNOM, did make the PnP for Polycom. Keven Moros even recommend Polycome for conference phone. That before PNXnSIP was acquired by SNOM.
PNP are available but their is certainly a bug their.
the licence we are using is (C:\Program Files\pbxnsip\PBX)
Please advise.
In regards of FreeBSD... I am very sorry, but it look something like no details to the little elements are in place. To me it look like if this is their, who know what at other places that we do not know... Sorry to be so attentive to details.
Many thanks again
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I don't think snom ever added the PnP support for IP6000.
Regarding the "freeBSD" on the Mac OS build, that can be ignored. It was just a wrong text. It will not impact the functionality.
Many thanks for your reply,
PBXnSIP, now SNOM, did make the PnP for Polycom. Keven Moros even recommend Polycome for conference phone. That before PNXnSIP was acquired by SNOM.
PNP are available but their is certainly a bug their.
the licence we are using is (C:\Program Files\pbxnsip\PBX)
Please advise.
In regards of FreeBSD... I am very sorry, but it look something like no details to the little elements are in place. To me it look like if this is their, who know what at other places that we do not know... Sorry to be so attentive to details.
Many thanks again
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I am trying to auto provision Polycom IP6000 with all proper settings
Mac ID OK
PNP OK
Network OK
Polycom auto provisioning is orking fine internally as long as SIP IP Replacement List and IP Routing List are empty.
In this situation, devices on the other side of NAT (outside of the local domain) are not providing audio.
To get device working fine outside internal network, SIP IP Replacement List and IP Routing List need to be set.
In that case, Polycom auto provisioning is not longer working...
It look like it is one or the other...
what am I missing?
SIP IP Replacement List: 10.0.1.200/24.202.225.XXX
IP Routing List: 10.0.1.1/255.255.255.0/10.0.1.200/0.0.0.0/0.0.0.0/24.202.225.XXX
PBXnSIP Version: 4.3.0.5021 (FreeBSD) - error here, it is running on OSX machine and it say's FreeBSD...
If SNOM can trade my polycom IP6000 (just received and NEW) for a MeetingPoint. I am more then willing to trade it :-)
Many thanks for your help
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Can you give us a PCAP?
I have identified the problem without beiing able to solve it.
Version 4.0.1.3499 is doing something on DTMF signaliing with Mediatrix 1204 v5 (all) making the DTMF to other PBX not understanding DTMF for AA. The only setting that work but ONLY with another PBXnSIP will be :
voiciIfDtmfTransport : outOfBandUsignRtp
This will not work with other pbx ...
So if you PBXnSIP is usign Madiatrix 1204 v5 (all) stay away from PBXnSIP 4.0.1.3499 for now. I found no possible ways of getting it respecting all confirmity and Interoperabilities with mediatrix products.
PBXNSIP version 3.4.0.3201 is working fine in conjonctions of Mediatrix 1204
voiciIfDtmfTransport : inBand
I learned that in the very hard way...
Mediatrix frimware version: 1204 V5.0.27.222 SIP MIB 1.5.3.100
PBX support... what would you need, what can I provide you to get this fixed...
PCAP but How and what?
many thanks
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250ft is fine, 100 meters = 328ft. to extend after that just put any network switch, (with poe if needed) in the middle. There are also extenders avilable for sale, see http://www.ippbx.us/Home/tabid/408/txtSear...er/Default.aspx
Many thanks for the reply,
Strange, because the phone (SPA504G) refuse to register if cable is not shorter... may it be a limitation of the phone with cable lengh?
Really need a solutions here!
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In order to do a 500ft run you would need to put a switch/hub/repeater in the middle. Anything that will regenerate and retime the signal.
Can anyone recommend a suitable hardware to regenerate signal, what should I buy if I must be running for 250 ft?
Kind of urgent please
regards
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Since version 4.0.1 have been introduce (great version), Cisco phones that are using TLS and NOT UDP or TCP are no longer registering at all.
If I downgrade back to 3.4.X, Cisco Phones using TLS are working just fine
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Like all 4.0.X previous version, when calling from 4.0.X to a 3.4.X system, reaching AA. DTMF isn't working at all -- like if 4.0.X is not sending proper DTMF signal to 3.4.X
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4.0 changes the way certificates are being handled. Maybe you have to import the certificate again. If it still does not work, can you get a Wireshark trace?
Certificat?
We are not usign certificat with TLS and never used.
What setting do you want for SharkTrace?
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3.4.0.3201 system is running with Cisco phone, one particular is usign SIP settings TLS protocol Cisco SPA962 frimware V. 6.1.5 (a)
When usign PBXCTRL version 4.0.1.3475 the same Cisco phone is never registering. I have tested another Cisco phone, different model and different Frimware, same issus.
So I come to the conclusion that TLS on V 4.0.1.3475 hav a little something to prevent Cisco phone to work.
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Current build 4.0.1.3446 still not woring properlly with DTMF detections.
Outboud and in bound calls are not detecting DTMF with Mediatrix 1204.
Exacte same coniguration is working fine with version 3 of pbxnsip
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I should not comment at all...
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Is it just me that have the impression that PBXnSIP company and software is fadins out... Their is not mutch activities on the forum and not release since several mounts... V 4.0 hase not showed any public info since past may...
Hope I am mistaking
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Are you using plug and play?
I have tried with and without!
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Good day,
I am testing SNOM M3 with version 3.4.0.3102
Phone register and work just fine if registered over external Internet connection, but does not transport audio if registered on the same domain of the PBX.
Any idea on what can be wrong
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oh yes, i'm assuming snom.
if snom you will need to reset to default settings...i usually do from keypad...
from across wan not sure you'll be able to do ffrom http...hmmm
matt
From my experience, it depend of the phone itself. Linksys/Cisco yes you type http://xxx.xxx.xxx.xxx/advence/?reste
other phones... I have no clue.
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How well PBXnSIP current version is running under Vista 64 bits machine?
many thanks
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Does PBXnSIP is compatible with the following:
VoIP Ceiling Speaker (The CyberData SIP-enabled VoIP Speaker is a Power-over-Ethernet (PoE 802.3af) and Voice-over-IP (VoIP) public address loudspeaker)
and
VoIP Outdoor Intercom (The CyberData SIP-enabled VoIP Outdoor Intercom is a Power-over-Ethernet (PoE 802.3af) and Voice-over-IP (VoIP) door entry device that easily connects into existing local area networks (LANs) with a single cable connection. The intercom is compatible with most SIP-based IP PBX servers that comply with the SIP RFC. Its tamper-proof design allows the unit to be mounted securely and safely.)
many thanks
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What product can I use to create a door bell that with interact with PBXnSIP in 2 ways communication and then allowed me to remotely open the door.
cheapest possible! :-)
many thanks.
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Good day,
Juste a quick note, 4.0.0.3212 Win32 on our site is working fine but we cannot see call logs reports and/or Active calls.
Do we have anything new on that?
Does V4.0.0.xxxx Win32 newer then 4.0.0.3212 Win32 can be available for testing?
many thanks for the great job.
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me to i dont like the esupport system.
other companies are using it. same thing there.
searching either finds to much or nothing.
for example i am trying to find out what `Accept Redirect` on the trunk settings mean.
i search for `Accept Redirect trunk` the first result is `domain settings`
bad
One or the other, information is the exact same ... with esupport we lost exemples. A support site is not a definition dictionary like it is now but something that help you understand and figure thigns out. Exemples and "to do" and "to not do" are things that really helps... Not something like changing a system for an other with absolutelly nothing new on it.
Do you buy 2 book storage unit just in case the first one get too old... Same old, Same OLD, same OLD
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Sorry, 90 secs. :-)
Good day,
Juste a quick note, 4.0.0.3212 Win32 on our site is working fine but we cannot see call logs reports and/or Active calls.
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Look extremelly promessing
How to:
1- Set Max. number of concurrent registrations per extension: Other then availible choices 1,2,3,4 or 5,
current installation is set to 7 registrations / extensions
I can manually set it on the pbx.xml but it doies not reflect on the reg_settings page
2- Set Mailbox PIN digets: Other then availible choices, 1,2,3,4 or 5 ?
current system is set to 6 digets
Cannot find it on the pbx.xml
many thanks
Incoming anonymous calls behaviour
in Call Treatment
Posted
Thanks for your reply,