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gotvoip

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Everything posted by gotvoip

  1. Audiocodes has a SAS mode http://www.audiocodes.com/Content.aspx?voip=2990 where it sits in between the phones and the remote ipbx and if the hosted ipbx can't be reached the phones will route the calls directly out the analog lines. However when we try this with pbxnsip and snom phones it does not work. I attached a trace and a diagram. Any help will be appreciated. Not_WORKING_Internal_Hub.zip
  2. I added an entry in my personal address book and then I did a click to dial from there and I got a page not found.
  3. When the polycom's do pnp and there is a domain address book and multiple personal address books on the system they all get merged into the phones address book so phone A sees phone B's personal address book.
  4. I thought it got it from the endpoint phone number setting. Also in the ip to tel routing table you can do some manipulation there. I think the AC also supports preferred Identity and the pbxnsip can set that on the trunk. Usually the PSTN service provider will put the correct from header on automatically to avoid any fictional caller ID in the switch. ITSP's is a different story!
  5. Do you ever want the call to go to vmail? What about call forward no answer or set a cell phone number to an extension so it will fork the call.
  6. I couldn't find the client on counterpaths site either. I tested it when it was in Beta about a year and a half ago and they must have given up on it. Try www.kapanga.com they have a wm5 client and they also support fax on it.
  7. I have a customer who can't get inband dtmf's working with viatalk. I have it set to yes and still nothing. Outbound work fine. The invite shows no out of band. Here is the logfile. Any ideas? I tried changing media settings on the trunk and it is using 711. Logfile Clear or Reload the log. [7] 2008/05/16 15:56:45: SIP Rx udp:216.246.73.186:5060: INVITE sip:xxxxxxxxx@192.168.30.15:5060;transport=udp;line=c4ca4238 SIP/2.0 Via: SIP/2.0/UDP 216.246.73.186:5060;branch=z9hG4bK3711432b;rport From: "xxxxxxxxxxxxxxxx@216.246.73.186>;tag=as6002f06d To: <sip:xxxxxxx@192.168.30.15:5060;transport=udp;line=c4ca4238> Contact: <sip:xxxxxx@216.246.73.186> Call-ID: 625703bd2e99896065d3113d0f5f4327@216.246.73.186 CSeq: 102 INVITE User-Agent: Viatalk SIP Max-Forwards: 70 Remote-Party-ID: "xxxxxxxxxxxx@216.246.73.186>;privacy=off;screen=no Date: Fri, 16 May 2008 20:03:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 210 v=0 o=root 1086 1086 IN IP4 216.246.73.186 s=session c=IN IP4 216.246.73.186 t=0 0 m=audio 16312 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [7] 2008/05/16 15:56:45: UDP: Opening socket on port 52378 [7] 2008/05/16 15:56:45: UDP: Opening socket on port 52379 [5] 2008/05/16 15:56:45: Identify trunk (line match) 1 [9] 2008/05/16 15:56:45: Resolve 28: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:56:45: Resolve 28: a udp 216.246.73.186 5060 [9] 2008/05/16 15:56:45: Resolve 28: udp 216.246.73.186 5060 [7] 2008/05/16 15:56:45: SIP Tx udp:216.246.73.186:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.246.73.186:5060;branch=z9hG4bK3711432b;rport=5060 From: "xxxxxxxxx@216.246.73.186>;tag=as6002f06d To: <sip:xxxxxxx@192.168.30.15:5060;transport=udp;line=c4ca4238>;tag=dcc2ade9bb Call-ID: 625703bd2e99896065d3113d0f5f4327@216.246.73.186 CSeq: 102 INVITE Content-Length: 0 [7] 2008/05/16 15:56:45: Set packet length to 20 [6] 2008/05/16 15:56:45: Sending RTP for 625703bd2e99896065d3113d0f5f4327@216.246.73.186#dcc2ade9bb to 216.246.73.186:16312 [5] 2008/05/16 15:56:45: Trunk Via Talk sends call to 100 [8] 2008/05/16 15:56:45: Play recordings/att1.wav space20 [7] 2008/05/16 15:56:45: Set packet length to 20 [9] 2008/05/16 15:56:45: Resolve 29: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:56:45: Resolve 29: a udp 216.246.73.186 5060 [9] 2008/05/16 15:56:45: Resolve 29: udp 216.246.73.186 5060 [7] 2008/05/16 15:56:45: SIP Tx udp:216.246.73.186:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 216.246.73.186:5060;branch=z9hG4bK3711432b;rport=5060 From: "xxxxxxxxxx@216.246.73.186>;tag=as6002f06d To: <sip:xxxxxxx@192.168.30.15:5060;transport=udp;line=c4ca4238>;tag=dcc2ade9bb Call-ID: 625703bd2e99896065d3113d0f5f4327@216.246.73.186 CSeq: 102 INVITE Contact: <sip:1xxxxxxx@192.168.30.15:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.10.2474 Content-Type: application/sdp Content-Length: 172 v=0 o=- 7333 7333 IN IP4 192.168.30.15 s=- c=IN IP4 192.168.30.15 t=0 0 m=audio 52378 RTP/AVP 0 8 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=ptime:20 a=sendrecv [9] 2008/05/16 15:56:45: Resolve 30: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:56:45: Resolve 30: a udp 216.246.73.186 5060 [9] 2008/05/16 15:56:45: Resolve 30: udp 216.246.73.186 5060 [7] 2008/05/16 15:56:45: SIP Tx udp:216.246.73.186:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 216.246.73.186:5060;branch=z9hG4bK3711432b;rport=5060 From: "xxxxxxx@216.246.73.186>;tag=as6002f06d To: <sip:xxxxxxxxx@192.168.30.15:5060;transport=udp;line=c4ca4238>;tag=dcc2ade9bb Call-ID: 625703bd2e99896065d3113d0f5f4327@216.246.73.186 CSeq: 102 INVITE Contact: <sip:xxxxxxx@192.168.30.15:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.10.2474 Content-Type: application/sdp Content-Length: 172 v=0 o=- 7333 7333 IN IP4 192.168.30.15 s=- c=IN IP4 192.168.30.15 t=0 0 m=audio 52378 RTP/AVP 0 8 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=ptime:20 a=sendrecv [7] 2008/05/16 15:56:45: SIP Rx udp:216.246.73.186:5060: ACK sip:xxxxxx2@192.168.30.15:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 216.246.73.186:5060;branch=z9hG4bK6ec74547;rport From: "xxxxxxx@216.246.73.186>;tag=as6002f06d To: <sip:xxxxxx@192.168.30.15:5060;transport=udp;line=c4ca4238>;tag=dcc2ade9bb Contact: <sip:xxxxxxx@216.246.73.186> Call-ID: 625703bd2e99896065d3113d0f5f4327@216.246.73.186 CSeq: 102 ACK User-Agent: Viatalk SIP Max-Forwards: 70 Remote-Party-ID: "xxxxxxxx@216.246.73.186>;privacy=off;screen=no Content-Length: 0 [7] 2008/05/16 15:56:45: SIP Rx udp:216.246.73.186:5060: ACK sip:xxxxxx2@192.168.30.15:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 216.246.73.186:5060;branch=z9hG4bK075c8ace;rport From: "xxxxxx@216.246.73.186>;tag=as6002f06d To: <sip:xxxxxx@192.168.30.15:5060;transport=udp;line=c4ca4238>;tag=dcc2ade9bb Contact: <sipxxxxx@216.246.73.186> Call-ID: 625703bd2e99896065d3113d0f5f4327@216.246.73.186 CSeq: 102 ACK User-Agent: Viatalk SIP Max-Forwards: 70 Remote-Party-ID: "xxxxxxxx@216.246.73.186>;privacy=off;screen=no Content-Length: 0 [9] 2008/05/16 15:56:45: Message repetition, packet dropped [9] 2008/05/16 15:56:45: DTMF: Power: 0 0 0 0 0 0 0 0 0 [9] 2008/05/16 15:56:45: Last message repeated 2 times [9] 2008/05/16 15:56:45: DTMF: Power: 0 0 2 0 0 0 0 0 1 [9] 2008/05/16 15:56:45: DTMF: Power: 0 0 0 0 2 0 0 0 0 [9] 2008/05/16 15:56:45: DTMF: Power: 2 0 2 0 0 0 0 0 0 [9] 2008/05/16 15:56:45: DTMF: Power: 2 0 0 0 0 0 0 0 0 [9] 2008/05/16 15:56:46: DTMF: Power: 0 0 2 0 0 0 0 1 0 [9] 2008/05/16 15:56:49: Resolve 31: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 15:56:49: Resolve 31: a udp 192.168.30.105 12052 [9] 2008/05/16 15:56:49: Resolve 31: udp 192.168.30.105 12052 [9] 2008/05/16 15:56:54: Resolve 32: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 15:56:54: Resolve 32: a udp 192.168.30.105 12052 [9] 2008/05/16 15:56:54: Resolve 32: udp 192.168.30.105 12052 [8] 2008/05/16 15:57:02: Play audio_en/aa_enter_extension_number.wav space10 audio_en/aa_dial_name_prompt.wav audio_en/bi_4.wav space20 [9] 2008/05/16 15:57:06: Resolve 33: url sip:richmond-3.vtnoc.net:5060 [9] 2008/05/16 15:57:06: Resolve 33: a udp richmond-3.vtnoc.net 5060 [9] 2008/05/16 15:57:06: Resolve 33: udp 216.246.73.186 5060 [8] 2008/05/16 15:57:06: Trunk 1 (Via Talk) has outbound proxy udp:216.246.73.186:5060 [9] 2008/05/16 15:57:06: Resolve 34: udp 216.246.73.186 5060 [8] 2008/05/16 15:57:06: Answer challenge with username 17326064002 [9] 2008/05/16 15:57:06: Resolve 35: udp 216.246.73.186 5060 udp:1 [9] 2008/05/16 15:57:06: Message repetition, packet dropped [5] 2008/05/16 15:57:06: SMTP: Timeout [8] 2008/05/16 15:57:10: Play audio_en/aa_enter_extension_number.wav space10 audio_en/aa_dial_name_prompt.wav audio_en/bi_4.wav space20 [8] 2008/05/16 15:57:11: DNS: Add dns_cname mail.friedomtech.com mail.friedomtech.com.netsolmail.net (ttl=60) [8] 2008/05/16 15:57:12: SMTP: Connect to 205.178.146.50:25 [9] 2008/05/16 15:57:12: Resolve 36: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:57:12: Resolve 36: a udp 216.246.73.186 5060 [9] 2008/05/16 15:57:12: Resolve 36: udp 216.246.73.186 5060 [9] 2008/05/16 15:57:13: Resolve 37: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 15:57:13: Resolve 37: a udp 192.168.30.105 12052 [9] 2008/05/16 15:57:13: Resolve 37: udp 192.168.30.105 12052 [8] 2008/05/16 15:57:19: Play audio_en/aa_enter_extension_number.wav space10 audio_en/aa_dial_name_prompt.wav audio_en/bi_4.wav space20 [7] 2008/05/16 15:57:24: SIP Rx udp:216.246.73.186:5060: BYE sip:xxxxxxx192.168.30.15:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 216.246.73.186:5060;branch=z9hG4bK60615544;rport From: "xxxxxxx@216.246.73.186>;tag=as6002f06d To: <sip:xxxxx@192.168.30.15:5060;transport=udp;line=c4ca4238>;tag=dcc2ade9bb Call-ID: 625703bd2e99896065d3113d0f5f4327@216.246.73.186 CSeq: 103 BYE User-Agent: Viatalk SIP Max-Forwards: 70 Remote-Party-ID: "xxxxxx@216.246.73.186>;privacy=off;screen=no Content-Length: 0 [9] 2008/05/16 15:57:24: Resolve 38: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:57:24: Resolve 38: a udp 216.246.73.186 5060 [9] 2008/05/16 15:57:24: Resolve 38: udp 216.246.73.186 5060 [7] 2008/05/16 15:57:24: SIP Tx udp:216.246.73.186:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 216.246.73.186:5060;branch=z9hG4bK60615544;rport=5060 From: "xxxx@216.246.73.186>;tag=as6002f06d To: <sipx@192.168.30.15:5060;transport=udp;line=c4ca4238>;tag=dcc2ade9bb Call-ID: 625703bd2e99896065d3113d0f5f4327@216.246.73.186 CSeq: 103 BYE Contact: <sip:1x@192.168.30.15:5060;transport=udp> User-Agent: pbxnsip-PBX/2.1.10.2474 RTP-RxStat: Dur=38,Pkt=7,Oct=1204,Underun=0 RTP-TxStat: Dur=38,Pkt=1893,Oct=325596 Content-Length: 0 [9] 2008/05/16 15:57:31: Resolve 39: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:57:31: Resolve 39: a udp 216.246.73.186 5060 [9] 2008/05/16 15:57:31: Resolve 39: udp 216.246.73.186 5060 [9] 2008/05/16 15:57:39: Resolve 40: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 15:57:39: Resolve 40: a udp 192.168.30.105 12052 [9] 2008/05/16 15:57:39: Resolve 40: udp 192.168.30.105 12052 [9] 2008/05/16 15:58:04: Resolve 41: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 15:58:04: Resolve 41: a udp 192.168.30.105 12052 [9] 2008/05/16 15:58:04: Resolve 41: udp 192.168.30.105 12052 [9] 2008/05/16 15:58:06: Resolve 42: url sip:richmond-3.vtnoc.net:5060 [9] 2008/05/16 15:58:06: Resolve 42: a udp richmond-3.vtnoc.net 5060 [9] 2008/05/16 15:58:06: Resolve 42: udp 216.246.73.186 5060 [8] 2008/05/16 15:58:06: Trunk 1 (Via Talk) has outbound proxy udp:216.246.73.186:5060 [9] 2008/05/16 15:58:06: Resolve 43: udp 216.246.73.186 5060 [8] 2008/05/16 15:58:06: Answer challenge with username 17326064002 [9] 2008/05/16 15:58:06: Resolve 44: udp 216.246.73.186 5060 udp:1 [9] 2008/05/16 15:58:06: Message repetition, packet dropped [9] 2008/05/16 15:58:08: Resolve 45: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:58:08: Resolve 45: a udp 216.246.73.186 5060 [9] 2008/05/16 15:58:08: Resolve 45: udp 216.246.73.186 5060 [9] 2008/05/16 15:58:31: Resolve 46: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 15:58:31: Resolve 46: a udp 192.168.30.105 12052 [9] 2008/05/16 15:58:31: Resolve 46: udp 192.168.30.105 12052 [9] 2008/05/16 15:58:31: Resolve 47: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:58:31: Resolve 47: a udp 216.246.73.186 5060 [9] 2008/05/16 15:58:31: Resolve 47: udp 216.246.73.186 5060 [9] 2008/05/16 15:58:32: Message repetition, packet dropped [9] 2008/05/16 15:58:56: Resolve 48: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 15:58:56: Resolve 48: a udp 192.168.30.105 12052 [9] 2008/05/16 15:58:56: Resolve 48: udp 192.168.30.105 12052 [9] 2008/05/16 15:59:06: Resolve 49: url sip:richmond-3.vtnoc.net:5060 [9] 2008/05/16 15:59:06: Resolve 49: a udp richmond-3.vtnoc.net 5060 [9] 2008/05/16 15:59:06: Resolve 49: udp 216.246.73.186 5060 [8] 2008/05/16 15:59:06: Trunk 1 (Via Talk) has outbound proxy udp:216.246.73.186:5060 [9] 2008/05/16 15:59:06: Resolve 50: udp 216.246.73.186 5060 [8] 2008/05/16 15:59:06: Answer challenge with username [9] 2008/05/16 15:59:06: Resolve 51: udp 216.246.73.186 5060 udp:1 [9] 2008/05/16 15:59:06: Message repetition, packet dropped [5] 2008/05/16 15:59:12: SMTP: Timeout [9] 2008/05/16 15:59:13: Resolve 52: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:59:13: Resolve 52: a udp 216.246.73.186 5060 [9] 2008/05/16 15:59:13: Resolve 52: udp 216.246.73.186 5060 ) [8] 2008/05/16 15:59:17: SMTP: Connect to 205.178.146.50:25 [9] 2008/05/16 15:59:20: Resolve 53: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 15:59:20: Resolve 53: a udp 192.168.30.105 12052 [9] 2008/05/16 15:59:20: Resolve 53: udp 192.168.30.105 12052 [9] 2008/05/16 15:59:32: Resolve 54: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 15:59:32: Resolve 54: a udp 216.246.73.186 5060 [9] 2008/05/16 15:59:32: Resolve 54: udp 216.246.73.186 5060 [9] 2008/05/16 15:59:48: Resolve 55: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 15:59:48: Resolve 55: a udp 192.168.30.105 12052 [9] 2008/05/16 15:59:48: Resolve 55: udp 192.168.30.105 12052 [9] 2008/05/16 16:00:06: Resolve 56: url sip:richmond-3.vtnoc.net:5060 [9] 2008/05/16 16:00:06: Resolve 56: a udp richmond-3.vtnoc.net 5060 [9] 2008/05/16 16:00:06: Resolve 56: udp 216.246.73.186 5060 [8] 2008/05/16 16:00:06: Trunk 1 (Via Talk) has outbound proxy udp:216.246.73.186:5060 [9] 2008/05/16 16:00:06: Resolve 57: udp 216.246.73.186 5060 [8] 2008/05/16 16:00:06: Answer challenge with username 17326064002 [9] 2008/05/16 16:00:06: Resolve 58: udp 216.246.73.186 5060 udp:1 [9] 2008/05/16 16:00:06: Message repetition, packet dropped [9] 2008/05/16 16:00:09: Resolve 59: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 16:00:09: Resolve 59: a udp 216.246.73.186 5060 [9] 2008/05/16 16:00:09: Resolve 59: udp 216.246.73.186 5060 [9] 2008/05/16 16:00:09: Message repetition, packet dropped [9] 2008/05/16 16:00:13: Resolve 60: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 16:00:13: Resolve 60: a udp 192.168.30.105 12052 [9] 2008/05/16 16:00:13: Resolve 60: udp 192.168.30.105 12052 [9] 2008/05/16 16:00:32: Resolve 61: aaaa udp 216.246.73.186 5060 [9] 2008/05/16 16:00:32: Resolve 61: a udp 216.246.73.186 5060 [9] 2008/05/16 16:00:32: Resolve 61: udp 216.246.73.186 5060 [9] 2008/05/16 16:00:36: Resolve 62: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 16:00:36: Resolve 62: a udp 192.168.30.105 12052 [9] 2008/05/16 16:00:36: Resolve 62: udp 192.168.30.105 12052 [9] 2008/05/16 16:00:59: Resolve 63: aaaa udp 192.168.30.105 12052 [9] 2008/05/16 16:00:59: Resolve 63: a udp 192.168.30.105 12052 [9] 2008/05/16 16:00:59: Resolve 63: udp 192.168.30.105 12052 Copyright © 2005-2008 pbxnsip Inc. All rights reserved. See the license agreement for more information.
  8. If you go into phones and modems do you see the pbxnsip tapi service provider installed? When I run ethereal I see xtapi messages so that is a good place to look. I have it working on my system.
  9. Can you turn on the pstn logging and include it in a post for some clues?
  10. I think the issues is on outbound calls on a trunk you can specify the DID but you can't specify the friendly name. You can grab it off the extension but what if a company wants to always send "my company 978-555-1212" so the company is sent and not Joe Blow 978-555-1212. There is no way to specify this and if you screw around with the extension name then the dial by name is not applicable. Some swithches and service providers will pass the friendly name.
  11. If you want to use an internal card then the Audiocodes TP260 cards work great. For external gateways Audiocodes M1K is nice as is the Vegastream, Quintum, or Patton Gateways. The Sangoma cards with echo cancellor are ready for beta if someone would like to try that.
  12. Since upgrading our first site to PBXnSIP 2.0, we have noticed an undesirable change in the treatment of the calling party ANI. We need to see if there is a way to get it returned to the previous behavior. In the 1.5 release, an Account that is in a Call Foreward state, would deliver the call to the Forwarded destination and present the originator's (from the first call leg) ANI. In release 2.0, the same scenario, the call delivers the Account's DN, instead of the originator's ANI. Documentation on Rel. 2.0 makes note of changes in this area to support forwarding calls to a mobile instrument, however it does not say how to get back to the original Release 1.5 behavior. Please advise
  13. Can you route from trunk to trunk. A customer wants to have two pbx’s tied together in 2 countries and be able to dial out to locale numbers. So PBX-A has a trunk set up to PBX-b and P to A. They would like to dial ext numbers on PBX-b(no problem) and would also like dial a 6 plus 10 digits for local numbers this is where I need some help.
  14. Did you see the new feature where the pbx sends out a daily cdr report? You can actually cut and paste for that into excel. Would that do the trick?
  15. If you press 5 it will give you the envelope information. I think is 1. the vmail picks up and plays the greeting to the cell phone mailbox and by the time that finishes the pbxnsip message is getting recorded. 2. would be a nice addition to the feature.
  16. You mean the mwi still goes to the forwarded phone? A trace would be nice from the pbx. In regards to the dnd button how about programming the dnd button to send the star code to the pbx so the dnd is always set there.
  17. My understanding is that service flag is only to redirect the cell phone after hours so it doesnt ring say at 2am if someone calls the work extension. So it sounds like you are not seeing the service flag take effect after hours. The extension should always ring regardless of the service flag and only the cell phone should be affected by the flag.
  18. What kind of phone is it? Can you grab a trace? Is the phone running the latest firmware? An ethereal trace would be nice.
  19. Is dnd on in the pbx by any chance? clear the log and make a call do you see the invite hit the phone? the pbxnisp log? What does it say on the origination phone lcd? Is there an error messge like user busy or some other clues
  20. I am not sure what the problem is from reading the post. Did you start drinking already? Did you define the buttons in the buttons page and use pnp to download the config? How did you setup the buttons in the pbx? Were these wiki pages helpful http://wiki.pbxnsip.com/index.php/Assigning_Buttons http://wiki.pbxnsip.com/index.php/Snom
  21. My understanding is it will set the alert-info header in the sip header to a belcore defined ring tone that the phone should understand so they users can hear that it is different and must be coming from the hunt or agent group and not the extension. Did that work? I am pretty sure the snom's and polycom support it.
  22. Did you try to drop it into the tftp directory?
  23. Is this still happening? I would see that occasionally and I thought it was coming from them and when I redialed it worked.
  24. I know that it was added for the exchange server 2007 um integration so we could tell who to bill for the call on that trunk in the CDR. I thougth that if you look at the CDR it would show the AA was called first then transferred to the extension. What are you trying to do?
  25. gotvoip

    No hang up

    Here is an easy way to get a reorder tone so we can analyze it and figure out what the frequency and the duration is so we can configure the sipfxo driver to recongize the tone. 1. Set "Maximum VoiceMail Duration" to 30 seconds: Select localhost domain, under Settings, you will see "Maximum VoiceMail Duration". 2. Dial in from PSTN. When Auto Attendant answers, the extension that did not currently online, then you can leave message. For example, 45. 3. When you hear VoiceMail IVR and start to record, hang be up your PSTN phone (now the busy tone will be recorded). 4. Wait until the 30 seconds timeout, the line will be dropped. 5. Login as the extension 45 to check mail. (Select Settings -> Lists -> Mailbox) Listen to the voice mail and see if the busy tone is recorded. Right click on the voicemail and save the file to PC. 6. Send the recorded file to us.
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