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Rafeh Hulays

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Everything posted by Rafeh Hulays

  1. We want to support business customers. However, supporting alarm monitoring is a must. Have anyone had success in supporting alarm systems over VoIP using Vodia? What are the preferable settings.
  2. Hi, All the times on the voicemail stamps and call logs are WRONG. What is the ideal settings for: - The server native time (do we install it with GMT time) - In the PBXnSIP web interface Settings (/reg_settings.htm), there is a Timezone. What this should be set to? - In the PBXnSIP web interface admin domain Settings (dom_settings.htm), there us another TimeZone Regards, RAfeh
  3. I can register the PAC to the account. However, there are several problems: - You should not be able to monitor extensions except when allowed by that extension. Now you can monitor anyone. This is not good!!! But more importantly for now: While the extensions do show up on the list and you can see their state (DND or not!), when I call into the extension, I do not see who is calling and to that matter anyone calling. Other issue: - I think it is a mistake to have the sip passwd there and all of the different settings in the PAC. Rafeh Hulays
  4. One of our customer wants to use a client with GSM codec. Does PBXnSIP support GSM codec?
  5. The aito-attendand works as follow: message 1: You reached netfone services inc., please dial your extension. If you want to search our directory press 8 Message 2: please enter the name of the person you would like to call I can change message 1 but do I change message 2 to say: please enter the first name of the person you would like to call Rafeh
  6. Hi there, How can we get a trunk to display a company Name ID. We are able to enforce on the trunk ID the number. How would we enforce the Name ID. It is important I get an answer ASAP as we will lose the customer if we are unable to come with a solution promptly. Currently the name ID is displayed from the first and last name of the Account. We need to enforce a single company name: Theoretically, we can put: First name: Netfone - Last Name: Rafeh Hulays The problem arise when we have auto-attendant. The search feature works with the first and last name. If the caller in the above situation keys in Raf, then the call is routed correctly. If the caller keys in Hul, the call will not routed correctly. We have to have then the ability to explicitely ask the caller to key in the first name of the person. We do not have this ability at present. Rafeh Hulays
  7. Are there plan for call-back functionality on the Extension Account level?
  8. Hi, I think that there is a mis-understanding of the problem. The problem is that the ATA does not see the call at all. The phone does not ring at the ATA. It is not an issue of one way audio. This only happens on boxes where we migrated from 1.5 to 2.0. If we registers the device on a clean 2.0 PBXnSIP, it works as a charm. Something that may be related to this is that when we restart the PBXnSIP, we lose all the device registrations. Meanwhile if we do the same on a clean 2.0 install, this does not happen. This is causing us major headaches!!! This is not an issue with one ATA. We have had to move dozens of accounts to a PBX with a new 2.0 install but this is causing major headaches! Rafeh Hulays
  9. Version 1.xx had issues with devices behind NATs. Incoming calls were blocked by NAT and a caller hears silence before the voicemail answers. version 2.xx (new install) does not have this issue and we tested it with the device behind many NAT devices and it works. The problem we are facing is that we have several PBXes that started as 1.xxx and we upgraded to 2.xxx. The problem with the NAT remains. The same problem occurs for Susee and Debian flavours of the PBX. We are now using 2.1.7.2461 (Linux). A call from an outside number 778-893-9348 (cell) to a voice over IP line 604-637-0912. The end user device is registered with the PBX from behind a NAT. When the call comes in, the caller hears an extended silence and then get forwarded to the voicemail. The device does not see the call coming in. (Please note that when we use a new version 2.xx install, it works and the call rings on the end user device correctly). This is trace from a 2.1.7.2461 version of software that has been upgraded (multiple upgrades) from a 1.xxx version: SIP/2.0 200 Ok Via: SIP/2.0/UDP 208.68.18.226;branch=z9hG4bK5292.43508421.1 Via: SIP/2.0/UDP 159.18.161.67;branch=z9hG4bK5292.7a9b8681.0 Via: SIP/2.0/UDP 159.18.161.101:5060;rport=5060;branch=z9hG4bK-efaaaf4788145af9e2c8baffc0c7212e-159.18.161.101-1 Record-Route: <sip:208.68.18.226;lr=on;ftag=159.18.161.101+1+4fd41c+4a38210d> Record-Route: <sip:6046370912@159.18.161.67;ftag=159.18.161.101+1+4fd41c+4a38210d;lr=on> From: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;isup-oli=00;tag=159.18.161.101+1+4fd41c+4a38210d To: <sip:6046370912@208.68.18.226>;tag=030856d931 Call-ID: 66876E82@159.18.161.101 CSeq: 154088517 INVITE Contact: <sip:6046370912@208.68.18.228:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: netfone-PBX/2.1.7.2461 Content-Type: application/sdp Content-Length: 262 v=0 o=- 1325934170 1325934170 IN IP4 208.68.18.228 s=- c=IN IP4 208.68.18.228 t=0 0 m=audio 5782 RTP/AVP 18 0 101 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [9] 2008/04/25 07:53:46: SIP Rx udp:159.18.161.67:5060: ACK sip:6046370912@208.68.18.228:5060;transport=udp SIP/2.0 Record-Route: <sip:6046370912@159.18.161.67;ftag=159.18.161.101+1+4fd41c+4a38210d;lr=on> Via: SIP/2.0/UDP 159.18.161.67;branch=0 Via: SIP/2.0/UDP 159.18.161.101:5060;branch=z9hG4bK-37fd8013b7ae6f3f13c353abfbc326eb-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info Max-Forwards: 69 Call-ID: 66876E82@159.18.161.101 From: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=159.18.161.101+1+4fd41c+4a38210d;isup-oli=00 To: <sip:6046370912@208.68.18.226>;tag=030856d931 CSeq: 154088517 ACK Contact: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;isup-oli=00 Organization: MetaSwitch Content-Length: 0 [9] 2008/04/25 07:53:46: SIP Tr udp:207.6.229.160:60778: CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482 To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca> Call-ID: 5dc9a996@pbx CSeq: 27432 CANCEL Max-Forwards: 70 Content-Length: 0 [9] 2008/04/25 07:53:51: Last message repeated 3 times [9] 2008/04/25 07:53:51: SIP Rx udp:159.18.161.67:5060: BYE sip:6046370912@208.68.18.228:5060;transport=udp SIP/2.0 Record-Route: <sip:6046370912@159.18.161.67;ftag=159.18.161.101+1+4fd41c+4a38210d;lr=on> Via: SIP/2.0/UDP 159.18.161.67;branch=z9hG4bK3392.ee170516.0 Via: SIP/2.0/UDP 159.18.161.101:5060;rport=5060;branch=z9hG4bK-64fdd954f60e668f7627d251a924542b-159.18.161.101-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info Max-Forwards: 69 Call-ID: 66876E82@159.18.161.101 From: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=159.18.161.101+1+4fd41c+4a38210d;isup-oli=00 To: <sip:6046370912@208.68.18.226>;tag=030856d931 CSeq: 154088518 BYE Organization: MetaSwitch Supported: 100rel Content-Length: 0 [9] 2008/04/25 07:53:51: Resolve 317750: aaaa udp 159.18.161.67 5060 [9] 2008/04/25 07:53:51: Resolve 317750: a udp 159.18.161.67 5060 [9] 2008/04/25 07:53:51: Resolve 317750: udp 159.18.161.67 5060 [9] 2008/04/25 07:53:51: SIP Tx udp:159.18.161.67:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 159.18.161.67;branch=z9hG4bK3392.ee170516.0 Via: SIP/2.0/UDP 159.18.161.101:5060;rport=5060;branch=z9hG4bK-64fdd954f60e668f7627d251a924542b-159.18.161.101-1 Record-Route: <sip:6046370912@159.18.161.67;ftag=159.18.161.101+1+4fd41c+4a38210d;lr=on> From: RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;isup-oli=00;tag=159.18.161.101+1+4fd41c+4a38210d To: <sip:6046370912@208.68.18.226>;tag=030856d931 Call-ID: 66876E82@159.18.161.101 CSeq: 154088518 BYE Contact: <sip:6046370912@208.68.18.228:5060;transport=udp> User-Agent: netfone-PBX/2.1.7.2461 RTP-RxStat: Dur=25,Pkt=259,Oct=8288,Underun=0 RTP-TxStat: Dur=5,Pkt=259,Oct=8288 Content-Length: 0 [7] 2008/04/25 07:53:51: Other Ports: 5 [7] 2008/04/25 07:53:51: Call Port: 020ec8ff@pbx#1390526460 [7] 2008/04/25 07:53:51: Call Port: 5dc9a996@pbx#1253982482 [7] 2008/04/25 07:53:51: Call Port: ab17b219@pbx#327669878 [7] 2008/04/25 07:53:51: Call Port: b9b50168-472333bb@24.87.11.54#fc2e71a790 [7] 2008/04/25 07:53:51: Call Port: fd537cda-bd2baccc@10.0.0.3#35a4289630 [9] 2008/04/25 07:53:51: Using outbound proxy sip:207.6.229.160:60778;transport=udp because of flow-label [9] 2008/04/25 07:53:51: Resolve 317751: url sip:207.6.229.160:60778;transport=udp [9] 2008/04/25 07:53:51: Resolve 317751: a udp 207.6.229.160 60778 [9] 2008/04/25 07:53:51: Resolve 317751: udp 207.6.229.160 60778 [9] 2008/04/25 07:53:51: SOAP: Store CDR in http://208.68.18.230/call_logging/call_log.2.0.2.php <env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://soap.com/pbx"><env:Body><sns:CDR><CallID>66876E82@159.18.161.101#030856d931</CallID><Type>mailbox</Type><Domain>office2.voipportal.ca</Domain><From>RAFEH AHMAD HUL <sip:7788939348@159.18.161.101:5060;transport=udp>;isup-oli=00</From><To>"Netfone Telesales" <sip:6046370912@office2.voipportal.ca></To><ToUser>6046370912@office2.voipportal.ca</ToUser><FromTrunk>van1_ser1</FromTrunk><TimeStart>1209135206</TimeStart><TimeEnd>1209135231</TimeEnd><StatisticsForward>0</StatisticsForward></sns:CDR></env:Body></env:Envelope> [9] 2008/04/25 07:53:51: SIP Tx udp:207.6.229.160:60778: NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c To: Netfone Telesales <sip:6046370912@office2.voipportal.ca> Call-ID: go2zi5t9@pbx CSeq: 31263 NOTIFY Max-Forwards: 70 Contact: <sip:208.68.18.228:5060;transport=udp> Event: message-summary Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 103 Messages-Waiting: no Message-Account: sip:6046370912@office2.voipportal.ca Voice-Message: 0/0 (0/0) [9] 2008/04/25 07:53:52: SIP Tr udp:207.6.229.160:60778: NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c To: Netfone Telesales <sip:6046370912@office2.voipportal.ca> Call-ID: go2zi5t9@pbx CSeq: 31263 NOTIFY Max-Forwards: 70 Contact: <sip:208.68.18.228:5060;transport=udp> Event: message-summary Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 103 Messages-Waiting: no Message-Account: sip:6046370912@office2.voipportal.ca Voice-Message: 0/0 (0/0) [9] 2008/04/25 07:53:52: Message repetition, packet dropped [8] 2008/04/25 07:53:52: SMTP: Connect to 64.40.101.128:25 [9] 2008/04/25 07:53:53: SIP Tr udp:207.6.229.160:60778: NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c To: Netfone Telesales <sip:6046370912@office2.voipportal.ca> Call-ID: go2zi5t9@pbx CSeq: 31263 NOTIFY Max-Forwards: 70 Contact: <sip:208.68.18.228:5060;transport=udp> Event: message-summary Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 103 Messages-Waiting: no Message-Account: sip:6046370912@office2.voipportal.ca Voice-Message: 0/0 (0/0) [9] 2008/04/25 07:53:53: Message repetition, packet dropped [9] 2008/04/25 07:53:53: SIP Tr udp:207.6.229.160:60778: CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482 To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca> Call-ID: 5dc9a996@pbx CSeq: 27432 CANCEL Max-Forwards: 70 Content-Length: 0 [5] 2008/04/25 07:53:54: Call 5dc9a996@pbx#1253982482: Last request not finished [9] 2008/04/25 07:53:54: Resolve 317753: udp 207.6.229.160 60778 [9] 2008/04/25 07:53:54: SIP Tx udp:207.6.229.160:60778: CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482 To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca> Call-ID: 5dc9a996@pbx CSeq: 27432 CANCEL Max-Forwards: 70 Content-Length: 0 [8] 2008/04/25 07:53:54: Hangup: Call 5dc9a996@pbx#1253982482 not found [9] 2008/04/25 07:53:54: SIP Tr udp:207.6.229.160:60778: CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482 To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca> Call-ID: 5dc9a996@pbx CSeq: 27432 CANCEL Max-Forwards: 70 Content-Length: 0 [9] 2008/04/25 07:53:55: SIP Tr udp:207.6.229.160:60778: NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c To: Netfone Telesales <sip:6046370912@office2.voipportal.ca> Call-ID: go2zi5t9@pbx CSeq: 31263 NOTIFY Max-Forwards: 70 Contact: <sip:208.68.18.228:5060;transport=udp> Event: message-summary Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 103 Messages-Waiting: no Message-Account: sip:6046370912@office2.voipportal.ca Voice-Message: 0/0 (0/0) [9] 2008/04/25 07:53:55: Message repetition, packet dropped [9] 2008/04/25 07:53:55: SIP Tr udp:207.6.229.160:60778: CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482 To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca> Call-ID: 5dc9a996@pbx CSeq: 27432 CANCEL Max-Forwards: 70 Content-Length: 0 [8] 2008/04/25 07:53:55: SMTP: Received 220 mail.netfone.ca ESMTP [8] 2008/04/25 07:53:55: SMTP: Received 250-mail.netfone.ca 250-AUTH=LOGIN 250-AUTH LOGIN 250-PIPELINING 250 8BITMIME [8] 2008/04/25 07:53:55: SMTP: Received 334 VXNlcm5hbWU6 [8] 2008/04/25 07:53:56: SMTP: Received 334 UGFzc3dvcmQ6 [8] 2008/04/25 07:53:56: SMTP: Received 235 go ahead [8] 2008/04/25 07:53:56: SMTP: Received 221 mail.netfone.ca [8] 2008/04/25 07:53:56: Sucessfully sent email to <rafehh@yahoo.com> [9] 2008/04/25 07:53:57: SIP Tr udp:207.6.229.160:60778: CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482 To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca> Call-ID: 5dc9a996@pbx CSeq: 27432 CANCEL Max-Forwards: 70 Content-Length: 0 [9] 2008/04/25 07:53:59: SIP Tr udp:207.6.229.160:60778: NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c To: Netfone Telesales <sip:6046370912@office2.voipportal.ca> Call-ID: go2zi5t9@pbx CSeq: 31263 NOTIFY Max-Forwards: 70 Contact: <sip:208.68.18.228:5060;transport=udp> Event: message-summary Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 103 Messages-Waiting: no Message-Account: sip:6046370912@office2.voipportal.ca Voice-Message: 0/0 (0/0) [9] 2008/04/25 07:54:01: SIP Tr udp:207.6.229.160:60778: CANCEL sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-b3f50baba35bb7402ef7f65faf86e366;rport From: "RAFEH AHMAD HUL" <sip:7788939348@159.18.161.101:5060;transport=udp>;tag=1253982482 To: "Netfone Telesales" <sip:6046370912@office2.voipportal.ca> Call-ID: 5dc9a996@pbx CSeq: 27432 CANCEL Max-Forwards: 70 Content-Length: 0 [9] 2008/04/25 07:54:03: Last message repeated 2 times [9] 2008/04/25 07:54:03: Message repetition, packet dropped [9] 2008/04/25 07:54:07: SIP Tr udp:207.6.229.160:60778: NOTIFY sip:6046370912@192.168.1.64:5060 SIP/2.0 Via: SIP/2.0/UDP 208.68.18.228:5060;branch=z9hG4bK-2fd19f9c4169e38aafab5772750a150a;rport From: Netfone Telesales <sip:6046370912@office2.voipportal.ca>;tag=4e6609627c To: Netfone Telesales <sip:6046370912@office2.voipportal.ca> Call-ID: go2zi5t9@pbx CSeq: 31263 NOTIFY Max-Forwards: 70 Contact: <sip:208.68.18.228:5060;transport=udp> Event: message-summary Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 103 Messages-Waiting: no Message-Account: sip:6046370912@office2.voipportal.ca Voice-Message: 0/0 (0/0)
  10. The dial plan in the last two releases is all mucked up. For example, if we have the following dial plan: Trunk Pattern carrier 1 011([0-9]*)@.* Not Allowed 1900([0-9]*)@.* carrier 2 ([0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9])@.* Deny 1900([0-9]*)@.* If a person dials 5196530114 that the call will route into carrier 2 trunk. That is not the case. It seems that it matches the 011 in the strings and attempts carrier 1. This was not the case before. If someone dials 1604519002, the 1900 is detected and the call is denied. This is causing us major headaches as we have upgraded to the latest builds. Please fix the situation ASAP or recommend a path. We are using the Debian and Suse latest builds.
  11. It is unlikely that a person wants to call in their own extension. However, is there a way to have it as an option. In our application, it is REALLY desirable to do so ... Many of those interested in Business hosted VoIP are entrepreneurs who are 1-3 people shops. It will not be user friendly to have another number that they call ... it will be a lot easier for them to call their voip extension. Also, the call will show on their account and call display and not on a general auto-attendant. Rafeh Hulays
  12. CELL-PHONE INTEGRATION PROPOSAL This proposal integrate two forms of call-backs. A regular account has cell phone integration option: 1. no cell phone integration 2. cell phone integration Direct 3. cell phone integration Call-Back The customer can choose to set his account to either of the above options Option 1: No cell Phone integration - that is trivial Option 2: Cell phone integration - Direct calling - a caller cals in into their extension - PBXnSIP detect the caller ID and if it is the cell phone, present the caller with a message to enter PIN - PIN correct: Would you like to check your VM or Place a call Option 3: Cell phone integration - CallBack - a caller calls in - PBXnSIP detect that the caller ID and if it is the cell phone number, it simply does not pick up - PBXnSIP call back the number after 2 sec and provide caller with the menu options: to check your voicemail click 1, to place an outbound call, click this. OTHER ACCOUNT TYPES [ LOWER PRIORITY] There are two other account types that I also suggest: Simple Call-back Account: - This will allow up to 10 numbers to be entered for call-back - This is not a Extension account - A user call, PBXnSIP does not pick up - PBXnSIP check the list of numbers and if one on the list for the account, it calls the number back - Allow caller to place an outbound call Simple Direct Calling Account: - This will allow up to 10 numbers to be entered - This is not a Extension account - A user calls in, PBXnSIP check the list of numbers and if one on the list for the account, it ask the caller for a PIN. - Allow caller to place an outbound call Rafeh Hulays
  13. I have been testing cell integration and i think that the way it is implemented is not proper. I believe that it should be tied to the customer extension account immediately. This mean that if the cell phone calls directly into the extension, it detects the caller ID and thus immediately takes the caller into the menu to check your VM, press 1. To place a call, press 2. I believe that using the auto-attendant is flawed. Also the call log for the extension account, does not show record of the outbound call that is made. This feature is very important for entrepreneurs and sales people and they do not want to have auto-attendant. They want one number! I do not want to create one domain for everyone of these single accounts and assign a seperate number for the auto-attendant. Rafeh Hulays
  14. Can we have the Debian image of the latest version of S/W to test When do we expect that 2.1 become production ready? Rafeh Hulays NetFone Services Inc www.netfone.ca
  15. Re: latest release of 2.0 When we call in from outside into an account voicemail, the system is unable to detect DTMF. The same is true for auto-attendant. Both are working on the 1.5 versions of the S/W that we had.
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