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YSJ3010

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Everything posted by YSJ3010

  1. I checked 4.1.0.4026 (Win32) and you are send 486 and i looks like its the best we can do with the polycom for now
  2. I sent you a email with the trace please advise Thank You
  3. Allow multiple ACD calls on agent - even if busy is on now when I'm on the phone and their are 3 callers in the cue only the next caller will ring on my phone the other calls are will not show up on the phone ( i only get the next caller when i am on the phone)
  4. Initial greetings if possible to have the phones ring when the customer is hearing the Initial greetings(now its start when the caller has finished Initial greetings
  5. Queue position announcement: when there are 5 calls in the cue and 2 agents and a call is by head of the queue and the a phone that its ringing and ignores the calls the caller will be put in the beginning of the Queue and the system will tell him you have 4 callers before you and he just got top the first before
  6. Agent Group Agent selection algorithm when its setup to "use preference from agent settings"the system looking on "Currently logged in agents" not on "All agents for this ACD (e.g., "41 42 43")" so whom ever longed in last will receive the calls the first
  7. I upgraded to 4.2.0.3966 (Win32) and below in the problems that i have Call Transfer ( Polycom) when trying to transfer a call from ext 2088 to ext 2089 not blind and when ringing i press Transfer again before the other side will receive the call when no audio (this problem is only if you transfer between 2 polycom phones
  8. can you please post your experience?
  9. please post a link for (.3966). for windows 32 and Linux (CentOS 5 32) Thank You
  10. Hi I Just tried it on pnxnsip version 4.2.0.3961 (Win32) and i had the same bug on the polycom phone
  11. Hi we have a data base that has all the domains(accounts) and you can search by phone number and with a click of a button it logs you in to the right server ( DNS lookup ) puts in the user name and password and brings you to the right account settings so its like one big box I would recommended pbxnsip for a big environment
  12. HI is there a way to have the system say this call might be recorded after the caller selects a option in the folder \PBX\audio_en bi_recording2 that says this call might be recorded Please advise how to enable it Thanks
  13. YSJ3010

    Bypass Media

    Hi is there a way to Bypass Media we are ruining pbxnsip and we need a way to control all the trunks from all the servers in one place it should me something simple to do i just cant find it <action application="set" data="bypass_media=true"/> Please advise
  14. This would be a problem with your sip trunking provider
  15. Yes 1.when a caller records a message and the mail box is set to send a email and delete the message the message light will stay on 2. if the call is directed to the main box visa a hunt Group and the ext wast a part of the hunt Group the red light will stay off
  16. Hi using version4.1.0.4026 and 4.2 when a call is using a hunt group the State on the PBX will show alerting even if the call is connected the problem is when using "Watch the calls" the ext the light will still blink on the all the phones for that call Please advise
  17. I just retried it now and it's working Thanks You
  18. Good idea (i cant use this idea for some of my customers since they have it directed to a mail box for Night settings ) almost every system i know of the IVR would restart. can you please add this in the next release Please advise
  19. it doesn't hang up it just says "this ext dos not exists" and blank (you can enter a ext number if you know it the system doesn't restart the IVR)
  20. HI i got it working by changing the config file # PCC Standard Format File # SIP_RCV_DET_HEADER_1="N" SIP_RCV_DET_HEADER_2="N" SIP_RCV_DET_HEADER_3="N" SIP_RCV_DET_HEADER_4="N" SIP_RCV_DET_HEADER_5="N" SIP_RCV_DET_HEADER_6="N" SIP_RCV_DET_HEADER_7="N" SIP_RCV_DET_HEADER_8="N"
  21. Hi if a caller selects a wrong option the system we say "this ext number does not not exist" and the message does not start again can you please fix this we have customers(online stores) complaining that there customers get lost and don't call again please advise if there is a work around for now Thanks
  22. Hi I trying 4.1.0.4026 and i cant find how to add credit for a whole domain(account) (one credit for all ext) Please advise Thanks
  23. Hi Here what I have found that the issue seems to be In the Header Tosip:011@test.sample.com (in this case 011 is the Hunt group)and the ext number is in Contact: <sip:201@205.558.70.71:5060;transport=udp The phone only looks on the To header How can i change the PBX should sent in a Hunt group the ext number in the TO field? please see the traces below Working Message Header Via: SIP/2.0/UDP 205.558.70.71:5060;branch=z9hG4bK-16aae05cd9bb9acd3bf472f5af043c39;rport From: <sip:8643607108@test.sample.com:5060;user=phone>;tag=27929 To: "Michael" <sip:201@test.sample.com> Call-ID: f6575798@pbx CSeq: 7423 INVITE Max-Forwards: 70 Contact: <sip:201@205.558.70.71:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: TESTAGENT/4.1.0.4026 Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 241 From IVR to Hunt Group:not working Session Initiation Protocol Request-Line: INVITE sip:201@74.255.68.90:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 205.558.70.71:5060;branch=z9hG4bK-1a30a25af71b090cbfb3ff9fbd1cdd17;rport From: <sip:6789780511@208.94.157.10:5060;user=phone>;tag=2499 To: <sip:011@test.sample.com> Call-ID: 9a062256@pbx CSeq: 23352 INVITE Max-Forwards: 70 Contact: <sip:201@205.558.70.71:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: TESTAGENT/4.1.0.4026 Content-Type: application/sdp Content-Length: 243 Message Body Please advise Thanks You
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