YSJ3010
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Posts posted by YSJ3010
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any update on this ?
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I got the reply from Polycom
it should be tcpIpApp.sntp.daylightSavings.start.date="8"
Pbxnsip please fix that {tz dst-start-dayp} should work on EST
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Hi
i tried version .3981 and i didn't work see trace below
|44.693 | INVITE SDP (g722 g711U g711A g729 telephone-ev...RTPType-101) |SIP From: "Test" <sip:101@XXXXXXXXXX.XXXXX.com To:<sip:*87010@XXXXXXXXXX.XXXXX.com;user=phone
| |(1029) ------------------> (5060) |
|44.708 | 100 Trying| |SIP Status
| |(1029) <------------------ (5060) |
|44.720 | 401 Authentication Required |SIP Status
| |(1029) <------------------ (5060) |
|44.746 | ACK | |SIP Request
| |(1029) ------------------> (5060) |
|44.747 | INVITE SDP (g722 g711U g711A g729 telephone-ev...RTPType-101) |SIP From: "Test" <sip:101@XXXXXXXXXX.XXXXX.com To:<sip:*87010@XXXXXXXXXX.XXXXX.com;user=phone
| |(1029) ------------------> (5060) |
|44.764 | 100 Trying| |SIP Status
| |(1029) <------------------ (5060) |
|45.101 | 404 Not Found |SIP Status
| |(1029) <------------------ (5060) |
|45.123 | ACK | |SIP Request
| |(1029) ------------------> (5060) |
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Hi
By polycom phones it needs to be entered like below i email polycom to confirm I will post when they get back to me
how can we setup what {tz dst-start-dayp} should be?
tcpIpApp.sntp.daylightSavings.enable="1"
tcpIpApp.sntp.daylightSavings.fixedDayEnable="0"
tcpIpApp.sntp.daylightSavings.start.month="3"
tcpIpApp.sntp.daylightSavings.start.date="12" --------------------{tz dst-start-dayp}
tcpIpApp.sntp.daylightSavings.start.time="2"
tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1"
Thanks
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below is the settings for the time for polycom phones
the problem is that the PBX doesn't reply anything {tz dst-start-dayp}
the work around is to enter tcpIpApp.sntp.daylightSavings.start.date="12"
please fix this
<SNTP tcpIpApp.sntp.resyncPeriod="86400" tcpIpApp.sntp.address="{tz ntp-adr}"
tcpIpApp.sntp.address.overrideDHCP="0" tcpIpApp.sntp.gmtOffset="{tz gmt-offset}"
tcpIpApp.sntp.gmtOffset.overrideDHCP="1" tcpIpApp.sntp.daylightSavings.enable="{tz
dst-enable 1}" tcpIpApp.sntp.daylightSavings.fixedDayEnable="{tz dst-fixed}"
tcpIpApp.sntp.daylightSavings.start.month="{tz dst-start-month}"
tcpIpApp.sntp.daylightSavings.start.date="{tz dst-start-dayp}"
tcpIpApp.sntp.daylightSavings.start.time="{tz dst-start-hour}"
tcpIpApp.sntp.daylightSavings.start.dayOfWeek="{tz dst-start-wday}"
tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="{tz dst-start-last}"
tcpIpApp.sntp.daylightSavings.stop.month="{tz dst-stop-month}"
tcpIpApp.sntp.daylightSavings.stop.date="{tz dst-stop-dayp}"
tcpIpApp.sntp.daylightSavings.stop.time="{tz dst-stop-hour}"
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="{tz dst-stop-wday}"
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="{tz dst-stop-last}"/>
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*81 Call Barge In is not working properly
example what is not working
ext 2000 is on the phone and has a 2nd caller on hold
ext 2050 dials *812000
ext 2050 will be connected to the caller that 2000 has on hold (not joining the call that 2000 is speaking to)
Please advise
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Well adding your own extension to the black list may not be that useful. But the feature black/white list the last caller. Maybe we can avoid blacklisting the own extension.
i tried to blacklist the last caller well the pbx stayed 101 (my ext number) has bin added to the black list (not the last caller)
please advise
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what is the actual problem - CDR? or call not being able to pickup?
call not being able to pickup with the button on the phone
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Now since the CDR problem is not sending the -
see the invite
|21.512 | INVITE SDP (g722 g711U g711A g729 telephone-ev...RTPType-101) |SIP From: "test" <sip:101@xxx.xxxxxxx.com To:<sip:*87010@xxx.xxxxxxx.com;user=phone
| |(1027) ------------------> (5060) |
|21.527 | 100 Trying| |SIP Status
| |(1027) <------------------ (5060) |
|21.538 | 401 Authentication Required |SIP Status
| |(1027) <------------------ (5060) |
|21.568 | ACK | |SIP Request
| |(1027) ------------------> (5060) |
|21.580 | INVITE SDP (g722 g711U g711A g729 telephone-ev...RTPType-101) |SIP From: "Test" <sip:101@xxx.xxxxxxx.com To:<sip:*87010@xxx.xxxxxxx.com;user=phone
| |(1027) ------------------> (5060) |
|21.594 | 100 Trying| |SIP Status
| |(1027) <------------------ (5060) |
|21.901 | 404 Not Found |SIP Status
| |(1027) <------------------ (5060) |
|21.923 | ACK | |SIP Request
| |(1027) ------------------> (5060) |
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Hi
when you are watching the calls for other ext (BLF) on your phone, when one of the other ext have a phone call it will blink on your phone with a option to pickup the call
when you press the button on next to the light that is blinking the phone sends to the PBX *87010 (010 is my main AA) the problem is since Version 4 i think
before the last update when the CDR was good read http://forum.pbxnsip...-number-called/ it didn't work as well, well the phone sent *87+718-718-7188 (the number who called )
and the pbx send back 404
Please fix this
Thanks
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Hi
Since Version 4.2.0.3973 when you check the live calls and when you check the CDR is doesn't show the number to what the caller called too is show to what group number is went to
Start From To Duration Trunk
2011/03/02 9:31P Cell Phone (718-222-2222@xx.xxxxxxxxx.com) 010@xx.xxxxxxxxx.com(101) 00:56 xxxxxxxxxxxxxx
010 is my AA
Thanks
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HI
when i try to black list a called after a call *92 the system stayed 101(my ext) 101 was added to the black list
Please fix it
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Yes
the sip provider added the rage
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HI
I would like to have every domains recording in a septate folder in the recording folder (sub folder for each domain and daily a new folder in that folder ) what to i need to enter by "Record Location:" so i can give ftp- access to the end user to the recordings
Thanks
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Hi
how can i add a rage of IP address is "Explicitly list addresses for inbound traffic:" 6.6.6.0/24 would that work ?
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Hi
I have in the DTMF Match List about 200 entries and when someone gets to the IVR Node the server will jump to 100% and ir will take about 25 sec until the ivr will continue
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just had it today again
You have a new voice mail message from XXXXXXXXXXX (45 seconds). You will find the message in the attachment of this email.
and the message is only about 15 sec and its missing the rest of the message
Please advise
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the message got cut off prematurely
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i don't register to the sip sever for each domain i do it only on localhost
do you have any other ways to limit the calls to 4?
I do this by limiting CO LINES in the trunk settings.
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Let me try to repeat. You received the voicemail in am email attachment and there it was 53 seconds, while on the file system it was only 15 seconds? Was the content the same? Was there something in the long voicemail after 15 seconds (no silence)?
Yes after 15 sec its the end of the file no silence
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I received a voice mail to my email in the email it stayed (53 seconds) and the recording was only about 15 sec it sounds like the voice mail was 53 and the recording wast competed recording
I checked in the recordings folder and it only had the same port of the message,
I had this problem 3 X already ,well wen i try to duplicate the problem i can't
please advise
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The PBX knows two types for calls: Objects and Legs. The call license counts objects while other features may look at the legs. So 5 calls in the license should be sufficient.
In the main domain settings i what to limit the customer to 4 calls (4 objects) can you please fix this
thanks
sip proxy is trying to register
in Trunk Setup
Posted
Hi
My trunk 29 is a sip proxy i see in the logs that its trying to register
on the GUI theirs no option of Expires: 3600 only when you check in txt
please fix this
[5] 2011/03/07 10:25:13:Registration on trunk 29 (Ticketing SYS 1) failed. Retry in 60 seconds
# Trunk 29 in domain xxxxxxxxxxxxxxx.xxxxxxxxxxxx.com
Name: Ticketing SYS 1
Type: proxy
To: sip
RegPass: ********
Direction: o
Disabled: false
Global: false
Display: 1002
RegAccount: 1002
RegRegistrar:
RegKeep:
RegUser: 1002
Icid:
Require:
OutboundProxy: xx.xxxxxxxx:5080
Ani:
DialExtension:
Prefix:
Trusted: false
AcceptRedirect: false
RfcRtp: false
Analog: false
SendEmail:
UseUuid: false
Ring180: false
Failover: never
Privacy: rpi
Glob:
RequestTimeout:
Codecs:
CodecLock: true
Expires: 3600
FromUser:
Tel: true
TranscodeDtmf: true
AssociatedAddresses:
InterOffice: false
DialPlan:
Colines:
DialogPermission: