Vodia PBX
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Hi! I think I was able to figure it out. Instead of using the IVRNode, I used the AA (IVR Tab), and I was able able to route the call to Voicemail prompt.
Okay...
From the IVR node, you should also be able to send the call to 8xxx, where xxx is the extension number. Anyway, AA is easier.
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how about the way i have it set up now, is it good? or i shouldnt set the same pref number even it uses the same trunk?
and , what do i have to be caution as far as putting which one first? for example i know i cant set 102 to be 1* and replacement 1* since then all calls starting with 1 will use pref 102
Apart from using the same preference number twice the dial plan looks okay to me.
Which one first only plays a role if a entry in the lower part of the dial plan should not rbe reached because of a entry in the upper part (defining the exception first and then handling other cases). But usually that is a feature - for example if you have a special trunk for calls to New Zealand put that first, and if it fails you go to a more expensive route that also works for New Zealand.
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I have exactly the same problem as above but I don't understand the resolution.
I have extensions from 101-150. My AA says press 1 for sales and 2 for service...
I have extension input=after 3 digit...
when someone calls in and presses 101 to go directly to ext. 101 they go to salesman...who is being inundated...
What specifically do you suggest as a work around?
If you put "1#" into the direct destinations of the AA it will wait for a timeout.
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maybe someone tell me how to i set up that when i dial a 10 digit number it will automaticly add a 1 in frotn of the number?
The trunks also have a setting that tell the PBX how to present a number. Depending on the version, that even works for inbound calls. YOu might have to upgrade to version 3.2 to get that working.
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I am looking for a way to not hardcode the ANI for each extension, however use some type of variable in the ANI field so that it can pull, and use the ANI from the invite message. I have situations when running the hosting version, and I have 1 PBXnSIP server registered to another as a trunk, and right now the only way to send caller ID on a per station basis of the remote system is to create a trunk and dialplan per each outbound caller ID, The problem is this method is slow, and an administrative nightmare.
Is using a prefix in the trunk an option? In R-O-W companies usually have a common company prefix and then append the extension number there.
The other thing is that you can tell the PBX not to change the To/From headers. I believe that is in the domain settings.
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I mean from IVR Node to Voice Mail of "8"+43 (Agent#43).
Where can I put the 843 in the IVR settings?
Ehh - you mean "IVR Node"? Or auto attendant IVR tab?
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I need help with setting up the following failover solution
i have 3 trunk's and each of those 3 have a 2nd trunk group which the providor gave us to use for failover
in the main trunk i have Failover Behavior on all error codes, request timeout 3
here is how i have the dial plan set up:
PREF 100 - Unassigned
PREF 101 - E911 SIPGW 1
PREF 102 - E911 SIPGW 2
PREF 104 VOICE TRADING SIPGW 1 PATTERN 011* REPLACEMENT 00*
PREF 105 VOICE TRADING SIPGW 2 PATTERN 011* REPLACEMENT 00*
PREF 107 VP SIPGW 1 PATTERN 1800* REPLACEMENT 1800*
PREF 107 VP SIPGW 1 PATTERN 1877* REPLACEMENT 1877*
PREF 108 VP SIPGW 2 PATTERN 1800* REPLACEMENT 1800*
PREF 108 VP SIPGW 2 PATTERN 1877* REPLACEMENT 1877*
Now the way this is set up, if the trunk on PREF 107 VP SIPGW 1 is down, will it failover to PREF 108 VP SIPGW 2?
and what should i do if i want to add another trunk (a diff termination) in case V SIPGW1 & VP SIPGW2 goes down?
do i add lets say TSG SIPGW1 trunk and set PREF 109 TSG SIPGW1 PATTERN 1800* REPLACEMENT 1800* ?
and PREF 109 TSG SIPGW2 PATTERN 1877* REPLACEMENT 1877* ?
and do i need to enable failover on the trunk VP SIPGW2 on all error codes?
The purpose on the "all error codes" is that the PBX can differentiate between the gateway itself being busy or the destination itself busy (talking). In SIP the gateway is supposed to send a 5xx code if it has no more channels available; but there are gateways out there which send 486 (destination is busy/talking). If you gateway does it right and sends a 5xx code when all gateway channels are occupied, you should select that it fails over only on 5xx calls. If it is one of those buggy ones, then you have to failover on all codes.
The other failover is timeout. You can control with the timeout setting on the trunk how many seconds the PBX should wait before it generates a 408 response. This will also trigger the failover case as if the gateway would send a 408 code.
When the failover happens, the PBX resumes the processing of the dial plan with the next higher number in the plan. You should avoid using the same preference number in the dialplan as it makes this processing kind of random. When the PBX resumes the processing, it will again look for matches. The next line that matches will be executed then. This line can also failover and the game continues until either there is a result (e.g. the call connects) or the dial plan has no more lines.
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But my question is how can I activate a mailbox and link this mailbox to the IVR?
You mean putting a "8" in front of the extension, so that the call goes directly to voicemail?
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I found out what it was. I had my domain name in the setting set as the FQD name but in the sip header I was sending the ip address. so I added the alias of localhost to the domain and it works like a charm.
Yea, that is a good old problem. If you have only one domain, make sure that you have the name "localhost" either as primary name or as one of the alias.
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sadly I'm using v2. I guess I just create an extension with the numbers and forward the extension over to the auto attendent
No, on 2.1 you can put the names into two fields. In version 2 we had "primary name" and "alias" names. The primary name could be "010" and the alias names "9787462777 9787462778" (seperated by space).
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Generally speaking, if you want to use the PBX service, the PBX must have a routable IP address. Behind a firewall which does NAT that is not so easy. Actually, it is the intention of the firewall to make it as hard as possible.
Even if you get it working, it will likely be instable. Not being able to change the firewall does not make the job easier. And one thing is also guaranteed: It will be a lot of work, setting it up and keeping it running.
Bottom line: Try to make your life easier and change the setup. Ideally just get a public IP directly to the PBX and then it will be a easy setup.
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What things should I consider before doing this?
I would make a backup of the working directory and copy it to the new host. There are no differences between Windows and Linux, even the CRLF are the same. Of course, you need to have another executable for the PBX itself; but that should be easy in Linux.
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I have an auto attendant that will have 3 phone numbers all pointing towards it. I've got the first one in as a tel:NUMBEHERE and it works fine. How do I add two more aliases with additional tel: numbers?
In version 3.1, just use the field "Account Number(s):". Use a space to seperate the list elements.
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the 7960 doesn't have a hold or transfer button.
it was a soft button before
I believe hold was pressing the button right from the display (the one associated with the call). Transfer should show up as a soft key. Not sure if Cisco cancelled transfer as a feature. Maybe that depends on the firmware version.
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I have a problem with voice mail not being completly deleted. I delete the voice mail after listening to it but the system still thinks it is there. The MWI still flashes. I log into the actual accounts to try and delete them to no avail. I have attached a sample account. Look at the date and time of the first 5 which cannot be deleted. I am running version 3.0.1.3023
Mailbox of 3012
Time Number Duration Flags
2106/02/06 01:28:16
2106/02/06 01:28:16
2106/02/06 01:28:16
2106/02/06 01:28:16
2106/02/06 01:28:16
2009/01/23 10:29:39 18192757360 00:36
2009/01/23 11:26:28 16139374795 00:30
I believe that problem is solved in 3.1 version (or later). Maybe after the next upgrade check if the problem perists.
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I see the global settings for mysql cdr writing, and during a recent call about a pending installation I thought I heard the idea existing to create a daily CDR file that would mirror the Simple CDR settings. We plan to move all XML CDR, recordings daily, we plan to outpulse via TCP/IP the simple CDR output to a SMDR reporting system, and the creation of a daily CSV file that mirrors simple CDR output would be the Holy Grail.
There is also a way to write the CDR to a regular file. Similar to writing a log file, but only for CDR. You can specify the CDR filename in the SOAP CDR URL setting like "file:cdr.txt".
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I already came across this document, that is where i got the Use of "SIP IP Replacement List" from. From your answer i assume that there is nothing else to know, works perfect according to the document ???
I can only recommend to get a routable IP address (so called "public IP address"). All other things are dirty workarounds, and they are difficult to setup and difficult to keep them going. Ask you service provider for a IPv6 address (I know he would not give you one, but increase priority for the IPv6 project - nothing happens without customers asking for it).
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I have some 7960 phones, everything is working fine, but i would like to have a hold button, or transfer button? Or monitoring anyone else when on extension?
any one have any ideas?
also, anyone have any luck configuring the 7914 expansion?
Hold and transfer should work using the phone's buttons. Monitoring someone else will be tricky... The Cisco SIP implementation is a little bit "unclear" on this point.
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What conference phone would anyone recommend? That works well with pbxnsip ?
I would say we are talking about Polycom Soundstation IP 4000 here.
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When i try to register a phone with a full tel number 5147899234 i get a 404 error. when I use a short name 513 it works. I try to register the long number on a CS425/410 it works great. Any ideas if I am doing something wrong.
Do you have a country code set? Maybe try to register +15147899234. It could be a problem with the representation of the number.
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Next week i have to implement PBXNSIP with SNOM phones on a Windows 2008 Server (2 nic's) and a NAT connection to the VOIP provider. After doing reading about problems with NAT is seems that use of "SIP IP Replacement List" is the way to go in this setup. As it is the first NAT imlementation i will be performing i would like to know if there are any problems i might have. And if there is anything else i might forget to do besides creating Port Forwarding rules in the router for SIP and RTP.
Thanks for any advise....
Check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses.
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In previous versions this could also happen when PBXNSIP is trying to send an e-mail but failing to..... (misconfiguration).
Question for PBXNSIP: Is this stil possible in the current versions ?
I don't think so. Email-stuff is now running in a seperate thread.
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..that did not work. internal calls are fine, but dialing out there is no audio. Either direction. So i put the SIP replacement field and the other one back, now the polycom works internal, but still the same dial out problem. Again, Snom works fine.
Your network sugestion, that is basically what I have. One interface has a 1 to 1 NAT setup, which is for external access and SIP Provider connection, and the other interface is for internal only.
Dont quite know where to go with this next, like I said before, Snom 360 works fine. btw, is my IP Routing List syntax correct?
What router are you using? Does it support transparent DMZ? If that is the case you can pretty much use http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses.
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After reading into this further i found that there is supposed to be a global seeting "offer pickup" that can be turned on or off but i cannot find this setting...i have seen it before but do not know where i is! any help would be greatly appreciated!
There is a setting called "offer_pickup" (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File). If it is set to "true" then the PBX does offer pickup functionality instead of speed dial. IMHO it is not good, because it implies a race condition where instead of calling a party you pick a party's incoming call up.
Failed Web Login Crash
in General Setup
Posted
We'll soon come out with a version 3.2, which does fix the problem.