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Posts posted by Vodia PBX
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Okay, seems that was really a bug. The timeout for pressing a key kicked in in the 2nd auto attendant.
We'll put it into a 2.1.7 version. IMHO not a reason to release it yet, we are still collecting more things.
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Do you know if multicast works with Polycoms? I have about 20 extension in a paging group (unicast) triggered from a linksys pap2t in a nursing home - lots of paging.
I am not aware of multicast support from Polycom. 20 is a lot. Are overhead speakers an option?
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Did you turn the flag in the extension on that tell the PBX to "Send email on status changes"?
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PBXnSIP, have you ever noticed this situation, and is there anything that can be done to lower the utilization?
Sure, unicast paging is a CPU- and bandwidth-killer. That's why we offer also multicast paging.
I would say keep the unicast groups smaller than 10 extensions, preferrably smaller than 5. Otherwise you are just stressing the system so much that ongoing calls might suffer.
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(1[0-9]{10})@.*|([0-9]{10})@.*
This pattern has the problem that the 2nd pattern group will be in the replacement parameter \2. You probably want "(1[0-9]{10}|[0-9]{10})@.*".
(1?[0-9]{10}@.*There is a closing bracket missing.
You can also try 1xxxxxxxxxxx|xxxxxxxxxxx.
ERE are fun. If possible stay with the simplified patterns...
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I see two things that could be the problem here:
1. There is a nother process occupying the SIP port. Use "netstat -abn" to find out which process is listening on port 5060. Use the task manager to find out what process is using the PID that you see in netstat.
2. For some reason, the domain name "localhost" is not there any more and the domain "192.168.26.130" does not exist. Then the PBX rejects the request.
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Yes, OCS side of things are working fine, it just when using the "assume call comes from" option for calls between states that is an issue.
Yea. The problem is that the OCS has it's own way here... For non-OCS trunks you of course would not set it. At the moment it is something that just cannot be changed. At least I don't know how.
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Did you change anything? Are you using a ITSP? Maybe they changed something?
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Well, it can happen that one way is RTP passthrough and the other side is codec-aware. No reason for concern.
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Did you see the Wiki article on OCS? http://wiki.pbxnsip.com/index.php/Office_C...ications_Server should help you to get the mediation server working. The rest should be no big issue.
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The call is rejected because my ITSP reads that as an invalid TN. i.e. a billable TN of record within their switch.
Well, then the ITSP may want to see the number in a different way. What about trying out the few modes (RFC 3325, None, Remote-Party-ID)? Usually one of them gives usable results.
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Question: Do you use direct trunks between the two offices? Then the OCS-related settings should have no effect.
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"If the parameter of the calling extension is set, the PBX uses that parameter as the Caller-ID. Having this as the first option makes it always possible to override the following steps. However, usually using this parameter is not necessary."
No, this is the "Parameter 2". It is in the registrations tab of the extension. Using the tel: alias is problematic because then different users cannot share the same outgoing identity.
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Are you saying that 20-30 calls can happen in or out at the same time because of that virtual status?
At what point would I need to purchase another trunk if at all?
That is correct, a trunk does not have a call limitation. You need another trunk only if you want to talk to another destination, for example another PSTN gateway or another SIP provider.
The trunk is currently registered with callcentric. Can you tell me more about gateway mode and point to a setup plan for both? Since I am no way a expert I hope there is a detailed source for that?Check the Wiki - http://wiki.pbxnsip.com/index.php/Domain_A...stration#Trunks
Also maybe what would be the best way to go since I will be adding more departments soon? The call attendant piece is where I am stuck. I need to have different ones for each department with different general mailboxes.Since I have 100 user capacity. It would seem I could split that up into 5 parts so 1-20 is one department, 21-41 is the next and so on. But partitioning like that is a HOW?? It would seem that all external calls would be to the same number and the extensions would be the separating factor.
If I wanted a different external number to dial then I would have to buy another trunk. Is that correct?
If you are using callcentric, I would definitevely have one trunk per domain. That trunk will be used for inbound and outbound calls. That keeps things simple. They all have their own DID number, and you even get bills for every domain. The cost for the SIP trunks is not outrageous, and IMHO it is worth it.
So when my ADMIN is sitting at her computer she will be able to see the incoming calls on her PC. I was looking for a program or utility that will give her a interface other than what she can see on her phone. Can you elaborate more on how that could work?Some phones support a "sidecar" (Polycom, snom) that can be used for monitoring extensions. We are also working on a "pbxnsip attendant console" (PAC), which is a piece of software that runs on the attendants PC - but this is only in testing phase yet.
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I spoke to soon. the update-rc.d doesnt work in SUSE but I was able to get all of the procedures in place but to no avail as the pbx did not start.
In SuSE, you can use "/sbin/chkconfig --add pbxnsip" to set the links automatically (pbxnsip being the file that you put in /etc/init.d).
After the process runs, you must configure the firewall to allow the ports that you want to use (e.g. 5060, 49152-65535).
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Since I have one trunk with 3 concurrent calls, my first thought was to buy another trunk to increase the concurrent call capacity. I am not sure if I would even need another trunk if my current trunk had say... 10 concurrent call capacity??
Fortunately, the trunks are "virtual", that means there is no limit on how many calls you can have on that trunk.
I already created new domains for each department (company) so that should work. Using the tel:# option in the alias should help incoming calls get to the right extension. I am just not sure about how to set up the trunk/domain relationship so outgoing goes well.I think an important question here is if you register the trunk or use the gateway mode. Depending on that, you might want to choose different strategies.
The last part is the office ADMIN being able to monitor everything?Everything stays in the domain. The admin can see what is going on in the domain - but not outside of the domain.
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This works, except it cuts off half of my message on AA#2.
That AA is on another system, right? The problem was that the AA#2 gets started with DMTF already coming in, and it happily picks that up?
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OUTBOUND PROXY ADDRESS WILL BE MYGATEWAY ADDRESS ?
Yes I think having the outbound proxy set on the trunk is extremly useful. You can just set it to the IP address of the other PBX, which is your destination.
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yes the entry to the dialplan part i was getting stuck if u could elabrate i would be eternally grateful
Priority: something low enough so that the other entries are processed later
Trunk: That would be "AtoB"
Pattern: E.g. 6141236543
Replacement: That can be left empty.
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Automatic Call Detail Reciepts would be nice to help trigger these charges no given accounts.
You can do that relatively easy with a IVR node. See http://wiki.pbxnsip.com/index.php/Linking_..._to_an_IVR_Node. The next version will make it possible to call an external number from a IVR node, that was possible yet. But you can already call an extension that has the cell phone forking enabled.
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Should be possible:
- Set up a trunk "AtoB" on PBX A that points to PBX B. Use Gateway mode for that and set the domain (e.g. b.com) and outbound proxy (e.g. sip:64.63.42.21).
- Set up a trunk "BtoA" on PBX B that points to PBX A, same as above.
- Add an entry to the dial plan on PBX A that routes calls with a speicifc prefix to trunk "AtoB". Make sure that the priority is higher that the other "default" entries. As pattern you can use the whole telephone number of the other office (e.g. 6144356542) or just a short prefix (e.g. 614*).
- Add an entry to the dial plan on PBX B that routes calls with a speicifc prefix to trunk "BtoA".
- Set up a trunk "AtoB" on PBX A that points to PBX B. Use Gateway mode for that and set the domain (e.g. b.com) and outbound proxy (e.g. sip:64.63.42.21).
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hi want to know the steps how i can do pbxnsip point to point
You mean you want to trunk from one PBX to another? So that you can call from location A to location B directly over the Internet?
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Also prior to allowing a caller to access your cell phone challenge them to agree to the terms of you service.
I think if you give callers the option to call the cell phone, they will do that. It will be difficult to charge them for that - one thing is how you automatically generate the bill, and the other question is if you have a business relationship with them at all. Plus do you really want to charge a customer because the PBX redirected the call to a cell phone? If I would be the customer, I would say why does that guy not sit in the office and pick up my calls?
But I agree: The cell phone integration topic is not over. This will be one of the hot topics over the next couple of years.We have to keep this on hte radar and make it very usable.
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Hmmm. Can you turn SIP logging on for the address 127.0.0.1? It would be good to see the INVITE coming from the PSTN gateway.
If you can reproduce the problem, chances are good that we can solve the riddle...
Remote extension
in Extension Setup
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Check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses. Bottom line: If you want something stable, get a public IP address...
Also http://wiki.pbxnsip.com/index.php/One-way_Audio might be interesting in this case.