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Kurt Harnish

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About Kurt Harnish

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  1. That really helps me. Great support as usual. Thank you!
  2. Yes and the area code. I posted the wrong version of Snom One. It's 2011-4.3.0.5020. For some reason the voicemail envelope is saying plus before the number.
  3. Is there any way to remove the Plus symbol in front of the caller id in version 5.2.3?
  4. I'm not sure if the active call setting is what we want but I will test it. I will post this to the SNOM phone forum as well since you are no longer SNOM?
  5. 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. 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
  7. We have a customer using 5.1.1 and when they record calls using the record button they show up on the web interface but when they use the retrieve button they are not there.
  8. 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
  9. Is it possible to listen to Agent Group recordings by calling in from the outside? I know it saves them to a folder but we need a way to listen remotely. Thank you, Kurt
  10. 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
  11. This error was on this morning Alarm [system.overload.cpu] : ALARM_ON
  12. It is showing that all of the channels are idle and ready.
  13. 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. 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.
  15. Is there any way for one extension to set the call forwarding on another extension via a star code or an IVR node? Thank you, Kurt
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