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Everything posted by grichardomi

  1. thanks for the follow-up. i dont know what "tapi driver" is. i'm simply an end user, not too technical, trying to install pbxnsip unto a window 2008 64 bit machine. as mentioned, i've installed this software successfully on 32 bit hardware. i'm baffled and stunned of lack of software stability on win 64. for example, i can successfully register one phone, come back 1 hour later and the site become unavailable or crashed. Then without any changes, it comes back to life 4 hours later. i'm sure someone had some success tweaking it, requiring more than just an automated installation.
  2. i'm convinced it's a win64 hardware issue - experiencing same with 32bit software version. or at the very least the pbx software should have made necessary hardware settings during installation. i'm running a remote virtual 2008 r2 server. any ideas anyone?
  3. i'm about to give up on pbx win64. is it possible to install win32 on windows 64 bit. if so what are the tricks?
  4. Looks like pbx is LISTENING on port 80 - how do i get around this problem?
  5. It's most likely a port conflict. This must be a win64 only issue, cause I've not had this problem on other win32's. Any idea on how to resolve this?
  6. Just downloaded latest windows 64 version. Have an Office10 license w/two trunks. It keeps crashing (This webpage is not available ...) when more than one phone extensions are added. Got messages while registering snom or asstra phone.
  7. Hello, I'm having same problem. Running Office10 plus version 3.4. Every pbx service started, CPU goes to 100%. Any idea what could be the problem? Tks, guy -
  8. I can make outbound calls ok. Thanks for the response.
  9. Everything worked fine until today. Phone rings and keeps ringing while it tell caller to leave message - or goes to voice mail. When its picked up, phone goes dead. I rebooted the phone with no success. Again nothing got changed in my environment.
  10. Below is my call log where BYE response is quickly dectected . Also I noticed for this "firstchoice" extension, Bind to Mac Address has nothing in it. But for localhost, all extensions have an "*" in the Bind to Mac Address column. When I tried placing an "*" for firstchoice domain, it would not let me. [2] 2007/12/01 03:08:56: Trunk status 2 (callcentric) changed to "200 Ok" (Refresh interval 30 seconds) [5] 2007/12/01 07:47:37: Identify trunk 2 [5] 2007/12/01 07:47:37: Trunk callcentric sends call to 600 [5] 2007/12/01 07:47:47: BYE Response: Terminate 200789-3405509249-139600@msw1 [5] 2007/12/01 07:48:20: Identify trunk 2 [5] 2007/12/01 07:48:20: Trunk callcentric sends call to 600 [5] 2007/12/01 07:48:28: BYE Response: Terminate 200864-3405509291-741536@msw1 [5] 2007/12/01 07:56:23: Identify trunk 2 [5] 2007/12/01 07:56:23: Trunk callcentric sends call to 600 [5] 2007/12/01 07:56:32: BYE Response: Terminate 201870-3405509774-920885@msw1
  11. Is it Aastra or server related problem? I beleive the problem is related to our 2nd, non-localhost domain. Our first localhost domain extensions are working fine. But anything extension attached to this new "firstchoice" domain does not work properly on aastra phone.
  12. Could not get permanent releif from this problem. When phone dies 4 seconds after it's picked up. The weird thing is, it seem to work when answered using over the ear headset. I'm pasting my phone local configuration below: upgrade file name: "480i CT.st" upgrade ip address: vendor: "Aastra Telecom" model: 480iCordless firmware md5: 25a1097ee901cefa054a9beb4e4b6543 softkey1 type: line softkey1 label: "line 5" softkey1 line: 5 dndkey value: 0 ringer volume: 8 time server disabled: 1 sip line1 auth name: 601 sip line1 password: 601 sip line1 user name: 601 sip line1 display name: 601 sip line1 screen name: "GUY RICHARD" sip line1 proxy ip: sip line1 proxy port: 5060 sip line1 registrar ip: sip line1 outbound proxy: sip line1 outbound proxy port: 5060 sip line1 dtmf method: 0 sip line5 auth name: 600 sip line5 password: 600 sip line5 user name: 600 sip line5 display name: 600 sip line5 screen name: "line 5" sip line5 proxy ip: firstchoice sip line5 proxy port: 5060 sip line5 registrar ip: firstchoice sip line5 outbound proxy: sip line5 outbound proxy port: 5060 sip line5 dtmf method: 0 handset list version: 2 key list version: 3 Feature key 10 En label: "Line 5" Feature key 10 Fr label: "Ligne 5" Feature key 10 Sp label: "Línea 5" Feature key 10 control: 2 Feature key 10 base event: 5 Feature key 11 En label: None Feature key 11 Fr label: Aucun Feature key 11 Sp label: Ningún Feature key 11 hs event: 0 Feature key 12 En label: None Feature key 12 Fr label: Aucun Feature key 12 Sp label: Ningún Feature key 12 hs event: 0 Feature key 13 En label: None Feature key 13 Fr label: Aucun Feature key 13 Sp label: Ningún Feature key 13 hs event: 0 Feature key 14 En label: None Feature key 14 Fr label: Aucun Feature key 14 Sp label: Ningún Feature key 14 hs event: 0 ftp server: ftp username: grichard ftp password: 0908y6
  13. thanks for the reply. I posted this question to our service provider on chat session please review this log and let me know if ok - Customer: i have one question: does your router to this server ensures qos , possible a SBC? Thomas S: No it does not do QOS Customer: what do you have that does? Thomas S: I don't think any dedicated server company in the world has a standard server for the cheapest price with QOS to a specific provider. QOS is very expensive hardware to provide. Thomas S: But we don't have any server with QOS Thomas S: But in a rack we can provide QOS hardware Customer: ok thanks Thomas S: But without QOS our routers are very fast
  14. I welcome suggestions or input on remore hosting with 13 extenstions on two domains. Any licensing issues? Below is the dedicated server option: Pentium 4 2.4GHz + 512MB RAM + 1x 80GB Drive + Windows Server 2003R2 Standard Edition + 5 IP Addresses + 750GB Monthly Transfer + 10mbps Uplink
  15. At this point, I'd like to hire a consultant to troubleshoot this problem. Anyone with info can email grichardomi@gmail or call me at 515.282.1455. Thanks
  16. Here my log while calling. I'm not sure if you can spot something unusual - [7] 2007/10/26 21:39:54: Other Ports: 1 [7] 2007/10/26 21:39:54: Call Port: 572060-3402439021-452147@msw2#80b5c21dda [8] 2007/10/26 21:39:54: Resolve destination 52829: url sip:d37fcde5091b5e9773a12e548f678fb2@;transport=udp [8] 2007/10/26 21:39:54: Resolve destination 52829: a udp 5060 [8] 2007/10/26 21:39:54: Resolve destination 52829: udp 5060 [8] 2007/10/26 21:39:54: Send Packet BYE [8] 2007/10/26 21:39:54: UDP: recvfrom receives ICMP message [5] 2007/10/26 21:39:54: BYE Response: Terminate 572060-3402439021-452147@msw2 [8] 2007/10/26 21:39:54: Resolve destination 52830: udp 5060 [8] 2007/10/26 21:39:54: Send Packet 200 [8] 2007/10/26 21:40:01: Resolve destination 52831: udp 5060 [8] 2007/10/26 21:40:01: Send Packet 200
  17. How do you perform a SIP trace? Here is some more info that might help locate the problem. I have 2 domains (localhost,firstchoice) The problem occurs on firstchoice domain extension. I've registered one phone for both. I have no user agents - only one extension account on firstchoice domain.
  18. Hello A call comes in, I say hello. the caller hears hello and that's it. However I hear my echo as I speak, but caller cannot hear me. Any suggestions?
  19. Just go this from Vonage which I hope would solve this problem: Vonage has identified a potential impact to some SoftPhone and SIP client applications due to network changes we are making effective October 16, 2007. If you are using a Vonage-supplied Soft Phone, please click here to download the latest firmware to ensure uninterrupted phone service. If you are using your own SIP client application, please note the following: Vonage performs extensive testing on Vonage-approved devices that we distribute and on the Vonage-supplied SoftPhone client. However, it is not possible for Vonage to be aware of or test all of the possible 3rd party software or devices that may be in use in conjunction with the SoftPhone lines. If you are using your own SIP client, Vonage recommends you download the Vonage-approved client by clicking here, or you can test your client prior to October 16, 2007 by using the following SIP proxy for testing purposes only: Group DNS: a.vonim.com Port: 10000
  20. I use this setting for Vonage Outbound Proxy: sphone.vopr.vonage.net:5061 It times out or fails every 24 hours. To get it registered again, I have to change Port Number like :sphone.vopr.vonage.net:5061. When this one times out, then it's sphone.vopr.vonage.net:5061 back again. Does anyone have any idea why this is happening? Thanks Guy
  21. Detlef, Thanks for the reply. My inquiry to Suppliers indicated that Gateway is not needed for our SIP trunking operation. So PSTN is only nescessary if you have analog devices. If anyone disagrees, please comment. Guy
  22. I should mention that I dont have local phone service or analog devides, it's all VOIP ! Is the Grandstream GXW-4108 a digital product? The presentation emphasized DIGITAL gateway. Thanks Guy
  23. Thanks for the quick response. The GWX4108 product one that I'm considering. I'd like to consider Audiocodes or others, but they are various models making it difficult to decide which one (digital) is suitable for my small operation. Can someone recommend a specific brand or model, one that's does not require too many tinkering to install? Guy
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