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YSJ3010

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Posts posted by YSJ3010

  1. Well, thats a SIP problem. How do you successfully disconnect a call that is not connected yet? We could send 487, right now we are sending 486. Not sure if this would make a big difference.

     

    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. 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

  3. 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

  4. We were not able to reproduce it in a slightly later version (.3966). But there seems to be no change in that area. Only difference is that one of the phone used is snom & the other one is Polycom 331 (just had only 1 Poly)

     

    please post a link for (.3966). for windows 32 and Linux (CentOS 5 32)

     

     

    Thank You

  5. as a work around:

    snom ONE built in star code will transfer call i understand.

    put call on hold and dial the * code for transfer.

     

    the whole idea is that non snom devices that are allowed are meant for things like doorphone, cameras, etc.

     

    I'm not saying I like this but is how it currently works.

    If you have a suggestion here: http://snomone.ideascale.com/

     

    Matt

     

     

     

    Hi

     

    I Just tried it on pnxnsip version 4.2.0.3961 (Win32) and i had the same bug on the polycom phone

  6. 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

  7. 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

  8. I've had recent issues with people trying to dial my extension or enter a conference room using cell phones. Both instances were with separate android phones. Do you know of any such known issues? Or is there a setting that needs to be modified in the android phone on how they send tones to the PBX?

     

    Thanks for any assistance.

    This would be a problem with your sip trunking provider

  9. 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

  10. You can always use the (Direct Destinations) in the AA options and redirect them to a operator for example.

     

    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

  11. Can you provide us with the initial invite to the system? I justed tested on our system and the AA Does not hang up even If I press the wrong extension or random dtmf.

    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)

  12. 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

  13. 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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