DWAyotte

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,

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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.

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.

Are there any other options for a moderator? So far I have found *1 to send an email about the call and *9 to end it. I can't find anything on the support site and want to know all the options the moderators have. thanks

not really sure no. what is the rule of thumb on rtp ports? The most concurrent calls I have had is 16.

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?

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,

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.

What is best practice on setting up users that are outside the office and want a hardphone? Is it possible to do so without a VPN and still maintain high enough security? Thanks.

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.

Meaning that being on DND doesn't make you unavailable in the Q, so you would still have calls routing to you? Why would that be?

If you look at the log I posted, it shows the agents that are on recovery. It also shows the agents that are not logged in, and finally it shows the agents that are available.

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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.

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.

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.

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.

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?

I just edited my post above, instead of creating a new one, but any help is appreciated. Thanks so much.

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.