Vodia PBX
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I am getting the same problem. Only on outgoing calls and the request timeout was appening after 3 minutes. In the call log there is no duration on all outgoing calls. I have restarted the cs410 after changing a setting for call end detection and suddenly the duration on outgoing calls is there and able to place calls for longer than 3 minutes. Will follow up if the problem append again.
Try turning polarity reversal detection off on the PSTN gateway settings. The PBX probably believes that the call is still ringing, but it is already connected.
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With our current phone system, while in night mode, calls get forwarded to a couple of phones in our office, but the original caller id from the external user does not show up. We would like to see the caller id after the redirection from the main reception.
Okay. So there is no PSTN gateway or ISP involved. Usually after a redirection the phone should show something like "9787462777 R:100" on the display (the R: means that that call was redirected from 100). But that only works if the phone actually went to a real phone, auto attendant does not count. So I guess that is where the problem is?
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Animation is ok but I want to rotate ads and control clicks which would mean some sort of java code or calling a URL
Ehh. Maybe use a Flash movie?
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hi
sorry now the calls arive in the pbxnsip but the calls dont go to outbound for example , my sip provider is sending me calls to my pbxnsip and my pbxnsip must redirect the calls to audiocodes , but now the calls dont doit that for example
sip:59170533006@2xxxxxxxxx:5060 SIP/2.0 <this is the real number but the pbx transform the call to this To: "300" <sip:300@localhost> so tha call never terminate how can i fax that i need to arive to pbxnsip whit the real number and redirec that mumber to the audiocodes to terminate well de pbxnsip must do the CDR
Oh you mean having calls coming from a trunk and going out to a trunk. That is actually possible if you set "Accept Redirect" and "Assume that call comes from user".
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In the UK it is possible to make cheap calls to mobiles by dialling a 08717549090, waiting for the person to start talking, and then dialling the international number (i.e. sending the outbound number as DTMF tones).
Is it possible to route calls to mobiles through this number?
Costs are as low as 6p per minute (as opposed to 12p per minute with most SIP providers).
Someone mentioned recently that cell phones have a feature that allows to put pause into the address book entries. That was a trick to get it working.
The PBX itself does not support two-stage dialling. The first stage SIP is completely different from the second stage which would be DTMF. But some PSTN gateways may be able to perform this task. From the PBX perspective, all it does is dial a SIP URI!
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Yes, but the main question was if the cs410 supports IPv6, and as I understand from your answer, it really doesn't. We would need a new kernel for the appliance, can you provide one somehow?
Today the answer is "no". Ipv6 works great on Windows and PC-based Linux and FreeBSD. But for the appliance it is still work to do. I wish that the chip vendor eventually provides an upgrade that includes this. We are not very good at compiling tool chains and kernels...
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Why does the Speed Dial Code have to be a *code?
It would be nice if i could make it a regular extension.
also what is CMC?
Otherwise it will be quiet messy for the PBX to route the call to the right destination (and difficult for the user to understand whats going on). Call it "tradition"...
CMC is "client matter code". That is a new topic coming up. We'll extend on this to group address book entries together to form a "client" and use it in the CDR and in the ACD. Stay tuned.
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any way to run diagnostics on this thing to make sure its all working?
If the file system full? Check the status page of use the shell and enter "df /". We tried that version and it even did the hangup detection in a nice way so that we can think about getting some nice sleep at night!
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I left all other fields at their default value, the only thing I filled in was the "Bind to specific IP address (IPv4):192.168.3.64" I noticed after the reboot, it still responded to it's other ip addresses both for SIP, as well as HTTP I checked the pbx.xml, and I saw <port_bind4>192.168.3.64</port_bind4>
Nono - that is only for multicast RTP (it is in the multicase section of the Ports).
For every port, you need to specify where to hind the port. The dedault is something like "80", which translates into "0.0.0.0:80 [::]:80" (unspecified IPv4 and IPv6 address). If you want to bind to a specific IP address, use something like "192.168.3.64:80". Those examples are for HTTP, you need to do the same thing for tje other ports with different port numbers.
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Is there a fix for this in the next release, please? It did not seem to be broken in the 2.x release.
The next build will force the same timezones for midnight reset and reporting.
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I am attempting to bind the PBXnSIP system to a specific IP so the other one is available to run IIS (as well as I do not want it to be able to respond to sip requests on the other IPs). I filled in the IPv4 address in the settings/ports/Bind to specific IP address (IPv4): field, however after a full reboot, the web, and sip ports respond to other IP addresses assigned to the system. Are there additional steps I need to take? I am running windows version 3.0.1.3023
Did you put something in like "198.133.219.25:80" (for HTTP)?
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is there a support for IPv6 in cs410 kernel? I cannot see ipv6 as a loaded module and there is no /proc/sys/net/ipv6 -directory. The kernel I'm running seems to be 2.6.11.7-1.08.5.1-IPX-series.
Yea, we also took a quick look at this. From the PBX application point of view, everything is compiled in. Maybe you are able to get a IPv6 interface set up, then the PBX should be able to use it.
comcerto:~# ifconfig eth0 inet6 add 2001:0db8:0:f101::1/64
No support for INET6 on this system.
On the compilation machine:
$ grep -i ipv6 kernel-linux_2.6.11.7-1.08.6_cs410/.config
CONFIG_IP_TCPDIAG_IPV6=y
CONFIG_IPV6=y
# CONFIG_IPV6_PRIVACY is not set
# CONFIG_IPV6_TUNNEL is not set
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i want to do a question..
I have a pstn line with 1 phone... that i want is when a customer call my phone a welcome message start play..
I dont want to buy a PBX because are to expensive...
i didn't find something at internet because i don't know how to search about it...
(Maybe i am not in the right section but i couldn't find a section about it...)
Well, the CS410 sounds like what you could use. If your budget it limited, you can try to compile open source software and run a PC with that software (keep in mind that some PC take 400 W, this is not "for free"!); the tradeoff here is that this cost you a lot of time time and the result may be a little bit unstable. But in any case you need something that connects to your PSTN line and you'll have to pay something for a FXO interface anyway.
Alternatively you can search for hosted services, where you don't run the PBX yourself; this job is done by a service provider for you. You don't have to worry about the technical details; just pay some monthly fee (not expensive!) and your are all set.
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Well, this is no problem with the phone. We also experienced the problem, but with an older version and testing it at the moment with a newer version. Maybe the problem can be solved by a software upgrade, we'll let you know.
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Does pbxnsip support H.281 for far end camera control? Is there a way we can do this on SIP? The customer is using standard video phones with pbxnsip.
Well, H.281 is part of the H.323 umbrealla. The name "pbx-n-sip" suggests that H.323 is not supported... So the answer is "no".
But you have another choice. Many modern cameras support SIP. For example, mobotix camera support SIP (www.mobotix.com).
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I think this is because the system time zone is not the domain time zone. When the system performs the mightnight events, it sets the timestamps for the agents in system time. Then when the statistics are being sent, the domain time zone is used. That explains the differences.
Workaround: Choose a system time that is the same as the domain time zone.
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How about the caller id feature? When in night mode, do we specify which phones to ring and does it show up with the actual caller id of the person?
Not sure if I understand correctly... You mean the ringback tone should annouce the redirection target? Should this number mixed into the audio stream, just like in the ACD case?
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So I can't offer them a solution for this. It's like you say. ITSP does not support callerid sending out, which was ofcourse the easiest for the receptionist she knows she has to pickup the phone with other greeting. But as this isn't planned on a short term I have to invent some other workaround for them.
If anyone have suggestions, please let me know.
The next version has a cell phone forking that may read out the number and asks the user to press "1" to answer the call. That might be a workaround.
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My browser says it is complaining about
java script://void(0)
Whow that helped. There was a stupid include of some code that was not used. It does not show this now any more, consider it fixed in the next version.
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One of the features that was requested by my collegues was when night mode is set, the handets ring a different tone to all users in the office. So the user is able to tell whether the call was forwarded (with the regular ring tone) and the night mode ring (another tone)
Not only that, they want to be able to see the caller id of the original caller when it's in night mode.
Are these features available?
Well, currently that is not possible... We have "vanity" ringback tones somewhere on the todo-list. Maybe if we address this topic we should also consider having them dependent on the time of day.
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When I tested the release, it worked fine on our test system (RHEL5). Or customer (Windows server) ver 3.0.1.3023 however, still have an issue with no stats in the email. Any ideas? One suggestion was a time zone difference. Customer is set to Eastern and server is in Central timezone.
Contents of email Report:
Activity report of agents:
Agent Name Availability Calls Duration (Hold)
103 Ann 0
128 Shel xxxxx 1193032:38-1193038:40 1193038:48-1193039:26 0
132 Luisxxxxxxxx 0
135 Marixxxxxxxa 1193031:51-1193039:45 0
138 Jxxxxxxxxxx 0
151 Mexxxxxxxx 00:00-24:00 0
Please wait one day/night. It could be that the last report is more than one day ago, and that would explain the strange availability times.
Was the starting time for Shel 16:38?
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I have a customer who has 8 Polycom 500 phones. We have taken them back to Factory settings updated the server logon info. Downloaded the Polycom 2.1.2 zip file into the tftp folder. When I go to register the phone they download the settings and I get a 0x20 error and phone just keep on trying to configure. One time I got to a point that the phone said configuraiton error 0x0.
We have been trying for two day now with no luck. HELP ME Please.
Is that Polycom 500 or 501? Is there a version 2.2.2 available for those phones? Maybe there is something wrong with the settings generated from hte PBX, which are based on firmware 2.2.2.
The fallback solution is to edit the PnP files manually and put them into the tftp directory. Then the PBX is just a plain TFTP file server.
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I upgraded to v3.0.1.2023 (win32). We run the interface secure using the self signed SSL certificate.
When navigating around the account listing pages there appears to be some element on the listing page - possibly and image - that is referencing non-SSL space.
This results in the browser message: "This page contains both secure and nonsecure items. Do you want to display the nonsecure items?" Yes/No.
Has anyone run into this same message? I imagine it would be something that pbxnsip has to fix in the code - looks like a reference to an image that has the http:// hardcoded so it will not load secure in SSL.
I have the same problem. But I just can't find what the insecure object is...
Recording multi-tenant
in Call Recording
Posted
Well, you can include the domain name in the recording path.