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DWAyotte

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Posts posted by DWAyotte

  1. Rind MOC and outbound calls via MOC

    in Microsoft OCS

    Posted

    I am trying to get communicator functioning completely.

    I want communicator to ring when a users extension is dialed as well as allow outbound calls to be made via MOC. For whatever reason I can't seem to get both working at the same time.

    Right now outbound calls via MOC are working properly, but communicator does not ring when the extension is dialed.

    I also noticed that the feature *00ext doesn't work, it seems to be because of the normalization rules do not like *, how can i modify my rule to allow for the * (for example to call VM *97)

     

    I have the checkbox within the mediation server to allow simultaneous ringing of phones.

    I have the proper registration in my account for MOC. +ext@mediation.domain.com;transport=tcp

     

    Any help is much appreciated. Thanks,

  3. Resize the PAC

    in PAC and WAC

    Posted

    You can not resize the PAC window. But for larger groups we are working on a monitoring page directly on the PBX web page.

     

    Is there any sort of API available or some method where we can create custom web pages with the information that we want to be on them?

     

    We want to setup a wallboard with the info we would like to present to our departments and then a webpage for supervisors to see all the info they need in 1 single page.

  4. Resize the PAC

    in PAC and WAC

    Posted

    Hey all,

     

    How can you resize the PAC window? We have supervisors that manage large groups under "Extension" and we need to be able to make the PAC window larger to make it easier to monitor all extensions. Thanks.

  7. Troubleshooting 1 way audio

    in General Setup

    Posted

    could you provide details on the configuration of the environment?

    audio issues are commonly related to NAT routers and routers no properly configured to dynamically manage SIP calls.

    On a PBXnSIP server with a Public IP exposed 1-way audio is less troublesome, but can be affected by adjacent devices on public IP's not playing fair..Internally on the LAN 1-way audio is less troublesome but LAN switches not properly configured for 802.1X features can result in the RTP streams being affected.

     

    If these calls are just with an external caller, and you are using a SIP provider, then look to any router that you may have in front of PBXnSIP? RTP ports may need forwarding on less smart routers.

     

    my pbx sits on my LAN and on the internet, no NAT (2 NICs). I have all the necessary internet facing ports open and the rest closed (a total of 100 RTP ports).

    I have had 1 way on LAN more times then I feel comfortable having.

    So if it is my switches, what can I do? What 802.1x features should I be verifying?

  8. Troubleshooting 1 way audio

    in General Setup

    Posted

    is it intermitting?

    try on the admin/setting/port page set Lock codec during conversation: to yes

    do the same for /admin/domain/selectyourdomain/trunks/ Lock codec during conversation: set it to yes

     

    This didn't seem to do the trick. I am still getting a handful of "I can't hear the caller" complaints each day. Any other ideas? I would love to be able to nail this down. Thanks,

  9. Out of office users

    in Other best practices

    Posted

    if you have the proper equipment, doing it via VPN is the best way to do it

     

    but yes, you can allow out of office users to connect to the PBX without a problem, as long as you don’t have port 5060 and rtp ports blocked on your PBX/firewall

     

    Without using a VPN, are the SIP Credentials passed in clear txt? Is there a way to do this via SSL or some sort of secure method? I have those ports open right now because I use a SIP Trunk.

  11. Troubleshooting 1 way audio

    in General Setup

    Posted

    I have frequent complaints about calls that only 1 party can hear the other. What is the best way to troubleshoot these sorts of issues? I have looked through a packet sniff, but am not seeing anything that looks to be an issue. Perhaps my eyes are not trained to see the proper errors. I am wanting to get some pointers from you guys on how to best troubleshoot these sorts of issues. Thanks a ton.

  15. Doesn't seem to be connecting

    in Agent Group Setup

    Posted

    I have an ongoing issue where my calls won't always connect to my agent groups. Here is 1 of many incidents.

     

    This comes from the pbxnsip log.

    [8] 20091013114014: ACD: Extension 232 is not logged in

    [8] 20091013114014: ACD: Extension 233 is not logged in

    [8] 20091013114014: ACD: Extension 245 is in recovery time

    [8] 20091013114014: ACD: Extension 274 is not logged in

    [8] 20091013114014: ACD: Extension 283 is not logged in

    [8] 20091013114014: ACD: Extension 285 is not logged in

    [8] 20091013114014: ACD: Extension 291 is not logged in

    [8] 20091013114014: ACD: Extension 301 is in recovery time

    [8] 20091013114014: ACD: Extension 306 is in recovery time

    [8] 20091013114014: ACD: Extension 311 has another call

    [8] 20091013114014: ACD: Extension 319 is in recovery time

    [8] 20091013114014: ACD: Extension 336 is in recovery time

    [7] 20091013114014: ACD: Next stage in 802 has 8 agents available, 1 selected

     

    Here you can see 8 Agents available.

     

    Wireshark reveals that the caller received this message: Status-Line: SIP/2.0 486 Busy

     

    The caller never was connected and none of the available agents phones rang. This issue is plaguing me. I appreciate any help on this.

  16. Sangoma support

    in Gateways

    Posted

    Hello,

     

    Thanks for the feedback...I"m not sure if pbxnsip works with 64bit...

     

    matt

     

    pbxnsip works in 64bit, I initially set mine up in 64bit and it worked fine. NBE does not, ok the driver is 32bit so the software can run, but since there is no driver there is no NBE. Sangoma didn't seem to be concerned at all when I inquired.

  17. Detailed Logs

    in SOAP

    Posted

    Is the SOAP method the only way to get detailed call logs? If so, I am comletely new to it and wouldnt know where to start. I just upgraded to the latest 3.3, does this version allow sending CDR's to an SQL DB? If so, are there any instructions on how to implement that? The SQL method would be great as long as we can extract enough information from the CDRs. Thanks a ton.

  18. Route to external number

    in Agent Group Setup

    Posted

    Hello all.

     

    I am trying to setup my agent group to failover to an external answering service of ours. I input the full number to route the Q caller to after 30 seconds, but when that timer goes off the caller receives an operator message stating the call is prohibited. How does one route a Q call to an external phone number properly? Thanks a ton.

  19. Close, but no cigar?

    in Microsoft OCS

    Posted

    Hi DWAyotte,

     

    please carefully compare your setup with the OCS pbxnsip wiki page: http://wiki.pbxnsip.com/index.php/Office_C...ications_Server !

     

    Trunk call: Could not identify user typically means you missed this part of the article:

     

     

     

    I am looking forward, to your feedback! ;) Good Luck!

     

    Best regards,

    Jan

     

     

    Jan,

     

    Thanks for the help, I had configured my "Assume that call comes from user" set to a registered user, but now that I changed that to a non registered user it seems to be working correctly.

     

    Now I have a strange issue where when I make a call and the caller picks up, my phone still rings. this happens both on my hardphone and through communicator. I assume a configuration issue on my AudioCodes MP114. But maybe you know more about this than I do?

  21. Close, but no cigar?

    in Microsoft OCS

    Posted

    I can't make any outbound calls via communicator or hardphones.

     

    When I dial on a hard phone the phone says "connected" and the call rings forever and ever and ever.

     

    If I make a call in communicator I get the Error 404 in communicator and error: Trunk call: Could not identify user - in pbxnsip.

     

    If I modify my OCS Trunk to send calls to an extension, then any call anyone makes with Communicator rings to this extension. I tried putting in the extension of my AudioCodes GW, which is 2, but that didn't render any result either.

     

    What am I missing? I am so very close, but I can't figure it out.

     

    Thanks.

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