Jump to content

grichardomi

Members
 View Profile  See their activity

  • Posts

    30

  • Joined

  • Last visited

 Content Type 

Posts posted by grichardomi

  1. Windows 64 Issues

    in TAPI

    Posted

    okay just to make sure: are you trying to run tapi driver or snom ONE itself?

    (i wasn't paying attention to what group this question was in and that you might be trying to run tapi driver--i was thinking snom ONE app earlier.;-)

     

    I know there are issues with running the tapi driver on a 64bit system.

    I'm not aware of any issues run 32bit snom ONE on 64bit Windows7 for example.

     

    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. Windows 64 Issues

    in TAPI

    Posted

    yes, you can run 32b snom one on 64b o/s.

     

    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. Windows 64 Issues

    in TAPI

    Posted

    Looks like pbx is LISTENING on port 80 - how do i get around this problem?

     

     

    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?

  5. Windows 64 Issues

    in TAPI

    Posted

    most often this is due to http or other ports already being used by windows/linux.

    Any chance http port being used by windows?

     

    snom:It sure would be nice if this error would be either logged in windows event log or or message pop up when snom ONe/pbxnsip starting. Even better would be to detect on install.

     

     

    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. Windows 64 Issues

    in TAPI

    Posted

    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.

  10. Drop Call on Remote Extension on Aastra 480iCT

    in Extension Setup

    Posted

    Usually those kind of problems come if people are not using an outbound proxy. Other problems are when phones place calls from received call lists where the contact is not clear. But it should be possible to locate the problem looking at the INVITE with the Request-URI.

     

    In your setting the line 1 looks a little bit suspicious because the domain name is a IP address. In multiple-domain environments it would be a little bit strange to have a alias name that is the IP address.

     

    Maybe you should try to temporarily remove the alias name "localhost" to really point out where the domain does not match in your setup.

     

    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. Drop Call on Remote Extension on Aastra 480iCT

    in Extension Setup

    Posted

    We found a problem when people are doing attended transfer and transfer the call before the other side picks up. Could this be a problem here?

     

     

    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. Drop Call on Remote Extension on Aastra 480iCT

    in Extension Setup

    Posted

    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

     

     

    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: 192.168.15.153

    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: 67.18.221.2

    sip line1 proxy port: 5060

    sip line1 registrar ip: 67.18.221.2

    sip line1 outbound proxy: 67.18.221.2

    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: 67.18.221.2

    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: 72.52.191.74

    ftp username: grichard

    ftp password: 0908y6

  13. Remote Hosting Option

    in General Setup

    Posted

    For 13 extensions that should be sufficient. Be careful with the IP addresses - one is enough, make sure the other four are not giving you unneccessary problems.

     

    For good user experience on audio, see http://www.pbxnsip.com/download/qos.ppt.

     

    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. Remote Hosting Option

    in General Setup

    Posted

    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

  16. Drop Call on Remote Extension on Aastra 480iCT

    in Extension Setup

    Posted

    Eehhm... Multiple domains are a complex topic. If this is your first installation, I would recommend to use only one domain and get some experience first.

     

    Also, check out http://wiki.pbxnsip.com/index.php/Log_Access for the logging issue. See www.wireshark.org for a tool that helps you find out whats going on.

     

    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@204.11.192.23:5060;transport=udp

    [8] 2007/10/26 21:39:54: Resolve destination 52829: a udp 204.11.192.23 5060

    [8] 2007/10/26 21:39:54: Resolve destination 52829: udp 204.11.192.23 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 192.168.15.1 5060

    [8] 2007/10/26 21:39:54: Send Packet 200

    [8] 2007/10/26 21:40:01: Resolve destination 52831: udp 192.168.15.1 5060

    [8] 2007/10/26 21:40:01: Send Packet 200

  17. Drop Call on Remote Extension on Aastra 480iCT

    in Extension Setup

    Posted

    I think then the best is to take a look at a Wireshark trace or at least at the SIP trace of the involved user agents.

     

     

    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.

  20. VONAGE Softphone - Outbound Proxy Timeout

    in Trunk Setup

    Posted

    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

     

     

    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

  21. VONAGE Softphone - Outbound Proxy Timeout

    in Trunk Setup

    Posted

    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

  22. PSTN: Is it a Must?

    in Gateways

    Posted

    oops, the GWX410x is pure analog PSTN... havent used any T1,E1,BRI VoIP products so I cant give you any help there.

     

    I am currently looking for an ISDN gateway into a phone system in Germany but that is not in my hands so I am not involved too much in the selection process.

     

    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

  24. PSTN: Is it a Must?

    in Gateways

    Posted

    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

  25. PSTN: Is it a Must?

    in Gateways

    Posted

    Well, for me I think its a safe backup in the case that our VoIP fails. Also I use the PSTN lines as dial-in numbers over a Grandstream GWX4108 gateway and for free local and toll-free calls, where our callcentric.com VoIP provider would charges us for. Also 911 is a consideration if the VoIP provider does not provide this function. Additionaly I need the PSTN for burglar and fire alarm anyway, so why not use it for some calls as well. Also our fax runs currently over a regular line - that way I didnt have to mess with the T.38 in the beginning of our VoIP rollout.

     

    We have about 20 extension each in two US and one Mexico location. Since we have Mexico involved and don't count on a stable internet connection - if this fails we use PSTN with a simple dialplan entry. For example if a user dials 99xxxx then the PBX uses PSTN. Due to the VoIP we cut the number of PSTN lines in half because most calls use now VoIP.

     

    So far we are happy with this solution. It safes us alot of money for calls but allows us to go back any time to call out on PSTN if we encounter problems, its not the cheapest setup looking at monthly cost but I think with the best functionality.

     

    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

×
×
  • Create New...