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Kurt Harnish

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Posts posted by Kurt Harnish

  5. Attended transfer issue

    in Call Treatment

    Posted

    I would rather not disable call waiting if possible. The call is coming in from the outside via a hunt group. When the are ready to transfer the call another call comes in and they press the transfer button twice it puts the attended transfer gets put back on hold and they lose the outside call from the hunt group.

  6. Attended transfer issue

    in Call Treatment

    Posted

    From the customer


    Call transfer – we have encountered this situation several times. We use the attended transfer feature where the receptionist puts

    a call on hold, presses a “one touch” key to dial the desired party in house, announces the call, and then presses the transfer button

    twice to complete the transfer. If another outside call starts ringing during this process, when Jolene hits the transfer button twice,

    the new incoming call gets disconnected and the call on hold that was supposed to be transferred remains on hold (is not transferred).


    SNOM ONE Version 5.1.3

    SNOM 720 with 8.7.3.25

  8. Shared mailbox

    in Mailbox

    Posted

    We have a customer running SNOM One v5. If I add extensions to the "Allow access for extensions:" or "Extensions that may access this mailbox:" it makes a copy of each message in the users mailbox. I didn't think the older versions of SNOM One worked that way. I would like it so it only stores the messages in the shared voicemail extension and if someone deletes the message it is gone from everyone's mailbox.

     

    Kurt

  10. Call Pickup Error

    in Call Treatment

    Posted

    Using 2011-4.2.1.4025 (Win64)

    This is the log when we try to use Call Pickup

     

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 192.168.1.83:2048;branch=z9hG4bK-s7vmz7xww33l;rport=43175;received=someipaddress

    From: "889" <sip:889@someipaddress>;tag=1129uy0e3n

    To: <sip:*87@someipaddress;user=phone>;tag=ea5a27f58a

    Call-ID: 3c2672a2cbc1-zaevthd57h5d

    CSeq: 2 INVITE

    Content-Length: 0

     

    [4] 2011/07/26 10:12:48: Ignoring directed pickup request for call leg because the difference of -882384765 is too short

    [7] 2011/07/26 10:12:48: Undirected call pickup failed

    [7] 2011/07/26 10:12:48: set_codecs: for 3c2672a2cbc1-zaevthd57h5d codecs "", codec_preference count 7

    [7] 2011/07/26 10:12:48: Set packet length to 20

    [0] 2011/07/26 10:12:48: SIP Tx udp:24.229.49.16:43175:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/UDP 192.168.1.83:2048;branch=z9hG4bK-s7vmz7xww33l;rport=43175;received=24.229.49.16

    From: "889" <sip:889@someipaddress>;tag=1129uy0e3n

    To: <sip:*87@someipaddress;user=phone>;tag=ea5a27f58a

    Call-ID: 3c2672a2cbc1-zaevthd57h5d

    CSeq: 2 INVITE

    Contact: <sip:889@10.1.255.1:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2011-4.2.1.4025

    Content-Length: 0

  13. Sangoma A101 Outgoing Call Failure

    in Trunk Setup

    Posted

    It is a PRI through Windstream. Incoming calls are working correctly but outgoing does not.

     

    Here is a debug from the Sangoma card.

     

    Codeset: 6

    DISPLAY 28 0b 4b 75 72 74 20 4c 61 6e 64 69 73

    2011-06-28 08:46:26:159 -0400 [5189:5341] INFO - netborder.pstn.sangoma.isdn.message.b1(B1 - A101_digital)di1(B1I1)-c23 : call-id=1309265186-61316-235745791-231 RECEIVING ISDN MESSAGE in LAPD PRIMITIVE DL_DA_IN via span 0

    RELEASE_COMPLETE

    Crv: 0x8001

    Codeset: 0

    CAUSE 08 04 82 ac 18 17

    Len: 4

    82: 1... .... : Extension indicator : Last octet

    .00. .... : Coding Standard : ITU-T standardized coding

    .... 0010 : Location : Public network serving the local user (LN)

    Cause: Requested circuit/channel not available

    2011-06-28 08:46:26:161 -0400 [5189:5254] INFO - netborder.pstn.sangoma.isdn.channel : call-id=1309265186-61316-235745791-231 [CALLING] No channel is available or the channel required as "exclusive" in previously sent SETUP message was refused. Event : connid=x3, chan=x17 : N_STATUS_INDICATION - Incoming call NO CHANNEL available!

    2011-06-28 08:46:26:161 -0400 [5189:5254] INFO - netborder.pstn.sangoma.isdn.channel : call-id=1309265186-61316-235745791-231 Make call operation failed because of reception of a DISCONNECT with ISDN cause=0x2c. No ressource are available for the outbound call.

    2011-06-28 08:46:26:162 -0400 [5189:5254] INFO - netborder.pstn.sangoma.isdn.channel : call-id=1309265186-61316-235745791-231 [CALLING] MakeCall operation failed. Cause=NO_RESOURCE_CONN_FAILURE

    2011-06-28 08:46:26:162 -0400 [5189:5254] INFO - netborder.pstn.BidirStateMachine.b1(B1 - A101_digital)di1(B1I1)-c23 : call-id=1309265186-61316-235745791-231 Failed to make outbound pstn call : All trunks are busy

    2011-06-28 08:46:26:163 -0400 [5189:5246] INFO - netborder.gw.CallLegWrapper : call-id=1309265186-61316-235745791-231 INLEG inviteRejected : NO_RESOURCE_CONN_FAILURE

    2011-06-28 08:46:26:164 -0400 [5189:5246] INFO - netborder.cdr : call-id=1309265186-61316-235745791-231 Call ended Tue Jun 28 08:46:26 2011

    2011-06-28 08:46:26:164 -0400 [5189:5253] DEBUG - netborder.voip.StateMachine.In : call-id=1309265186-61316-235745791-231 VoipStateMachine processing event eAPI_INVITE_REJECTED

    2011-06-28 08:46:26:166 -0400 [5189:5253] INFO - netborder.sip.message : call-id=1309265186-61316-235745791-231 SENDING SIP MESSAGE (RESPONSE) via UDP to 127.0.0.1:5060 :

    SIP/2.0 503 Service Unavailable

    Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f1f3ea6b2430264a0fc9501b7c9f1699;rport=5060

    From: "user" <sip:888@snomone.pbx.com;user=phone>;tag=868794664

    To: <sip:7177330793@127.0.0.1:7066;user=phone>;tag=ds-7f6af5df-3953f164

    Call-ID: 16a78289@pbx

    CSeq: 29047 INVITE

    Content-Length: 0

    Server: Netborder Express Gateway/4.1.1

    CPD-Result: ???

    Contact: <sip:127.0.0.1:7066;transport=udp>

     

     

    Thank you, Kurt

  14. Set forwarding on different extension

    in Call Treatment

    Posted

    One of our customers has a extension 771 (unregistered, just used for forwarding) that is a maintenance number that everyone knows. This number needs to be forwarded to different people throughout the day. On the old asterisk system they had 9001 9002 9003 setup that when they dialed any of those numbers it would forward that extension to different numbers. Ext 9001 would forward 771 to his cell phone 9001 would forward 771 to his home phone.

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