Parks
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Posts posted by Parks
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As long as you are able to split the user group up into domains with 75 calls per domain then you can do that today. You need to run several processes; each one bound to a seperate core (leave one core for the OS). Each process has a different working directory. And you either bind each process to another IP address or you use different ports. In Linux/BSD this is relatively straightforward. In Windows, you cannot use the standard installation procedure but must use another mechanism to set the services up.
In any case, the key is to be able to split users up into smaller pieces. Then you can scale along cores and servers.
That's a lot of work to accomplish this as you would need to separate everything including having separate web portals for each. This makes it increasingly difficult to manage. Do you have an estimate time line when this will automatically work making 1 login and binding all to 1 IP. This sounds like a duck tape job for now at least.
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I am wondering what version will be able to utilize multicore technologies so we can have more than 75-80 concurrent calls per server. We're wanting to be able to have at least 250+ before having to deploy more servers.
Also looking for in the future would be able to manage all domains from one management interface whether 2 or more servers. It would be nice to have the directory stored on a SAN that multiple servers can pull from and also making the service load balance.
Don't know if PBXnSIP is looking to do this kind of development or stay more towards the smaller ITSPs.
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I am looking for some assistance with getting this working. What player with best and what port is used. Also would like to provide different RTP streams for different domains. Thanks for the help.
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Got the answer. It's called Camp On which is under your domain setting in the middle of the page.
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How to disable that callback feature on an extension. This is only on some extension and we never enabled it and have done the same setup everytime. So I don't know where it comes into play. Please let me know where this can be turned off. Thanks
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not fixed for me either.
Me either and it's really bad as I don't want my cell being published. Hopefully it's fixed soon.
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Thank you.
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Can someone let me know how this is done? Thanks
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Do you have a wireshark? Can you check your firewall log?
After about 3 hours of testing we figured it out. We had other IPs that where on the NIC so our data center could ping all of our IPs per their requirements. For some reason pbxnsip binded incoming audio to the snom phone to that IP but all traffic wasn't suppose to be on that pictular IP. We removed and everything was back to normal.
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This morning we've experienced one way audio inward from callers. The callers can hear us but we cannot hear them. We've tested domain to domain, within the same domain and all has the same issue. We can call in to check voicemail fine. Call forwarding on extensions works also, which doesn't make sense. Calling in from cell to place call from pbxnsip also works fine.
Please HELP!!!
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Here is a trace from our carrier:
Session 237-1233077718@66.212.26.34, Release Cause (code) = 0402
T Stack Cause = 488, Not Acceptable Here
Start = Tue Jan 27 09:35:18 2009
Answer = NA
Release = Tue Jan 27 09:35:18 2009
Duration = 0
Post Dial Delay = 0
Ring Time = 0
Orig TID = 003061, Proc_SIP
ANI = 9515362807, From = 65.90.252.66
DNIS = 8886387810, To = 66.212.26.34
Call Leg ID = SD1io2601-77d00464efbb2926f4d5ad6cd8302e2c-v300gi1
Media IP:UDP = 64.192.8.82:10652
Switched IP:UDP = 66.212.26.34:10544
Codec List = G.711u_64k,G.729,RFC 2833
Term TID = 004059, Proc_SIP
ANI = 9515362807, From = 66.212.26.34
DNIS = 8886387810, To = 216.218.236.3
Call Leg ID = 237-1-1233077718@66.212.26.34
Media IP:UDP = 0.0.0.0:0
Switched IP:UDP = 66.212.26.34:10546
Codec List = NA
Final Call Indication = F
Orig: Pkt In = 0, Pkt Out = 0, Byte In = 0, Byte Out = 0
Term: Pkt In = 0, Pkt Out = 0, Byte In = 0, Byte Out = 0
Why would be getting this??? carrier sending traffic to interactive intelligence sip proxy which send to our soft switch.
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That's what I thought as well when I talked to our client. But what is the likelihood that everyone in that office that their phones are all bad. This doesn't happen to one or two users. It can be someone different everyday and normally last 3-5 minutes before they can make the calls again.
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That wasn't our case as it was ringing the cell immediately and our voicemail pickup was the default 20 seconds. We did update last night to a newer release and are fine now. I am also advising our clients if they want to use this feature to extend as you said the voicemail pickup to at least 30 seconds.
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Ok, I'll do this but when it happen today I saw in their domain that they were still registered but had no dial tone and couldn't make calls.
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I understand what you are saying but it would be PROACTIVE and responsible to have system that limit this possibility. If a customer has to be told "sorry you lost all your voicemails from today and your phones were down for 2 hours because he had a pissed off employee" you think they one care but more importantly they have our trust that we wouldn't allow full admin access to the system. Should be able to create other admin users where they can only view and create as I original said. This would make it possible to have employee access the system without fear of this event from ever happening.
This is not a problem for us now but as we grow I don't personally want to be responsible for every question or add.
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It's always after the prefix that it hangs up. This is happening to all domains and extensions that we've tested and or users tried. The system is forking the cell right away or after 5 seconds. So the the cell mailbox isn't picking up but the user is.
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Hi,
So on a few extension and on seperate domains when we have the forking on for someones cell and a caller calls in it rings the users cell fine and starts reading the caller ID but after the prefix the system hangs up entirely. No I know the caller goes to their voicemail still. Any one else having this problem. BTY, we 're running 3.1.1.3101
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We have a client that a mortgage company with about 12 agents and every so often users will not have a dial tone, but at the same time some other users on either on the phone talking just fine or making calls. Sometimes it's everyone and might be down for 3-5 minutes. Can tell if this is there internet connection or our system. Don't think it's us as other companies aren't calling in or emailing about it.
Any ideas?
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Hi All,
I don't know about other hosting companies but when I hire a employee I don't want to provide them complete access to the whole system. If they get pissed off they can single handlly cripple a company by doing multiple things to domains. It would be nice to add user but limit them certain access to areas and or duties like create, view only this way they cannot mess anything up either.
Please provide your thoughts on this topic and hopefully we can get this on the ToDo list ASAP.
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The version that you have probably has the NTP time server set (in the admin/ports page). Just clear it and then it should use the default time server again.
I'll try this tonight as it requires the services to be restarted.
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Hi. We have put the default time server in this field but after the phone reboots it goes back to our server for some reason. Can anyone think of why this is happening?
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That's what I thought the answer might be. Just checking however. Thanks for all the input.
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They do have an AA but only picks up if huntgroup doesn't answer the call within 20 seconds. The reciptionists are in that HG. Also this bunch aren't very techinal and don't really like using technology unless they have too. Any other ideas? Also what happens if more than one have the same PIN?
Multicore technology utilization - when will be available?
in PBX Lounge
Posted
Yeah, I think that would be a really good idea. I also still have amjor concern that I cannot add additional admin users and assign them specific roles to work on.