Vodia PBX
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Posts posted by Vodia PBX
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Thank you, now I know I am not going nuts. But what is happening to the voicemails that are being left? DOes PBXnSIP have a fix for this?
Can you get a WAV file and attach it here?
It could be the rollover counter of the SRTP. First guess.
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Does anyone know how this feature might apply to Nexvortex? They had never heard of it.
Check out http://www.911enable.com/. They have the documentation on how it works.
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Yea, it is amazing that web-based admin can scare professionals off. They prefer remote desktop when they can have HTTP.
Version 4 has a new web interface, designed by a web designer (not a C++ programmer
). Maybe that will help to present the buttons in a way that is easier to digest. -
Just would like those who've installed and use cs410 and have some experience let me know what issues I can expect with a cs410 in real life. ;-)
I'm "fiddling" with one but may need to put one at a client soon.
My advice:
Keep in mind this is a small embedded system. So: be careful with large hunt groups, transcoding, recording. In other words, use G.711 and keep groups small.
FXO is always a tricky part. Depending on carrier, caller-ID and hangup this will require some fiddling...
Well, and this is Linux. If you need to do anything advanced, get Putty ready.
And if you plan to run it on public IP, don't forget to change the default password also on the Linux.
We are using it every day.
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has anyone tried the Polycom VVX 1500 with pbxnsip yet?
Nope. I guess this will be pretty boring as version 3 does not fully support video yet. You first have to establish a audio connection before you can include video.
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Just to clarify and explain the process i want the call to flow is :
when the call comes in i want it to ring the receptionist at ext 300 during business hours. If a service flag 101 for night mode is on then i want it to call the hunt group at ext 102, which will ring all phones. in this hunt group the final destination will be the AA at ext 100, which will have the greetings based on the service flag 103. I noticed that this does work but the service flag 101 has to be set to "Set" in order for it to go to reception and when it is "Cleared" it goes to night mode. I think this is due to the hunt group having the service flag set to 101 with the number to 300.
Is there something wrong with this setup?
Well, you should think the way the PBX processes the request. If you want to ring the receiptionist during office hours, you would probably "hunt" her first (pretty small hunt group), and if the night mode is active send the call to the auto attendant (on holidays) or to another hunt group which rings the whole office.
What most customers do is that the hunt group for the receiptionist also rings other people in the office after some time. This covers the cases when the receiptionist is temporarily unavailable (everybody needs a break).
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Any plans to add the copy feature in upcoming versions? The time it is taking me is killing me as I am growing.
This is available, but only in a pretty unstable version 4...
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If you want to have a special greeting for the holidays, you should set up a service flag for the holidays and specify 00:00-24:00 for Monday-Sunday and list the holidays. That means "we are open anytime, except the holidays". For example, this might be the service flag "123".
The other service flag has the usualy 09:00A-5:00P settings for Monday-Friday. Lets call it "124".
Then put that service flag first, so that on any time during the holidays the caller will hear the holiday greeting. Then put the other regular service flag begind this flag, Service Flag Account="123 124" and set the redirection accordingly, Night Service Number="130 131" (130 for the holiday and 131 for regular days night mode).
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I'm attempting to setup an auto attendant to work alongside a peer proxy ring-group application..... with no success yet......
Call flow goes a little like this..........
itsp >>> proxy >>> pbxnsip autoattendant
caller dials an extenstion, call is routed back to proxy.....
proxy <<< pbxnsip
registered device (to proxy) rings
....... feature on proxy routes call back to pbxnsip if the call isn't answered......proxy >>> pbxnsip
call dies
get strange error messages:[5] 2009/04/14 12:13:15: Received loopback request without tag
[5] 2009/04/14 12:13:15: Received incoming call without trunk information and user has not been found
what header declares the 'trunk information'?
why is the user found when the same request URI is used (and the user is called directly)?
I guess you turned the loopback detection off already (admin settings).
Does the PBX received the request from the proxy as well? You can see that in the "Rx" in the log. If the request also comes from the proxy, then it should work...
Loops are a dangerous stuff. It can very quickly consume a lot of resources. That is why the PBX has a special settings for that. If you have some leisure, check out "Max-Breadth" in RFC 5393.
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Any word on this? I've been trying to setup this on my system for quite some time with no avail. I've read about changing the "1" to "0" and vice a versa, but nothing has worked. If someone has successfully created this sip.cfg or whatever file needs to be changed, could you post it here???
We did put that into the template. FYI it is only for the handsfree mode.
You can check if it made it into the config file by checking out the file in the "generated" directory.
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ok. in a multi domain enviroment when you pull a PNP config it always uses the ip of the NIC...
What do i use to override this with the domain IP? i assume there is a domain IP variable we can use instead in the template file. this will cause the cisco to work.
Multiple domains are just a problem with Cisco. You cannot use a domain name! You cannot use a domain name! You cannot use a domain name! I don't know what Cisco was thinking, but that is the way it is. Set up multiple domains, and set up multiple IP addresses, name each domain after the IP address.
We are hoping Cisco comes out with a firmware that supports DNS names in domains. Every crappy phone that I know supports this from day 1. It would be good for their reputation as a technology company.
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I have 4 Snom 820 phones (all on the same firmware) and after the latest PBXnSIP update, one of my phones has a dialing problem. When I enter the 4th digit of the number, the phone will auto-dial the 4 digits which are not a valid destination. My other three 820 phones don't do this and they all boot from the same domain and PBX. Any ideas on what's causing this - the phone is currently unusable as a result.
You probably have set it up for 3-digit dialing (this is a domain setting in the PnP setting area). You may dial numbers by starting with 1. For example, you you want to dial (978) 746-2777 then you would enter 19787462777.
If you don't want to have this behavior, you can change the PnP setting to "user must press enter". Then the user may use the phone like like he uses a cell phone. Enter the number, edit the number, and when done start the call with the green button.
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I'm currently using the Fax Pass Through mode, My SIP trunk provider (Teliax) does not support T38.
T.38 has been invented for a reason. The reason is that SIP trunking very easily looses packets. If you loose one packet when receiving a FAX, the rest of the page might become completely black (or white, if you are lucky).
I was on an older version of the CS410 and using the spa2001 and I had no issues. Unfortunately over the last six months I was battling a 4 second intermittent silence issue. That was recently resolved. It was a cable modem with ARP storm detection enables causing the issue. I still need to update my other 400 page post still. In that time, I upgraded to the latest software on the cs410, we changed the fax machine as the other one died, and I'm sure I have changed some settings on the PBX. All was well on the PBX side until we tried to fax. I worked with the SPA2001 for a week and reads horrible things. I upgraded to the ht286 and still no luck. Audio is crystal clear if I use the handset or a regular phoneo n the fax line. It is forced on g711u. We don't do much faxing and are ok with the occasional issue but I have tried 30 pages and not one has went through. Always a comm error. I hear the dee dee dee da shhhh and then i see the call online from the pbx for 60 more seconds or so. My guess it is some echo or silence supression issue. You mentioned this on another post but not sufre how to change on the cs410The CS410 does not change anything regarding echo compensation or silence suppression. Once the two calls are connected the PBX happily passes the audio packets from one side to another and that's it (unless you are performing transcoding). If the provider performs echo compensation on a FAX this might be an issue.
"One common problem with the Gradstream devices is that they advertize UPDATE, but when they receive a UPDATE message with a SDP offer, they don't answer this.Well, that is a major bug that they should really fix.
What you can do is to turn off the support for UPDATE. The setting is called "support_update" and is in the http://wiki.pbxnsip.com/index.php/Global_Configuration_File. I would try to set it to "false". ".Right, unfortunately in the SIP world Grandstream is not the only vendor with problems implementing UPDATE correctly... Check the pbx.xml file to see what the current value is.
Here is what Teliax says. I'm awaiting a reply from the support line. Is there anywhere in the CS410 to limit audio correction or clipping? They have an inbound fax service we will use for all inbound items.The only thing that I can think of is transcoding. But if you are using G.711 on both sides, the signal should not be changed at all. The PBX also does not perform VAD, and does not advertize this.
5. Can I use my fax machine?Yes. In most cases you can use a fax machine but this service is not 100% guaranteed. VoIP uses compression technology to preserve bandwidth and send a high quality voice signal accross the internet. Part of compressing a voice signal involves removing the higher and lower portions of the voice frequency being sent over the internet. These frequencies are inaudible to the human ear but are very important to machines like faxes. We have some special instructions for faxing over the TelIAX network. Please email us at support@teliax.com for more information.
Well, FAX is a problem in the VoIP world, this is not only a Teliax problem. T.38 does solve the problem of packet loss and it makes FAX possible over the regular Internet.
T.38 sounds like a solution that they should support, but the interoperability in T.38 is lously. And there is not even a way to encrypt T.38 packets like RTP and SRTP. T.38 does not even use RTP.
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Yes, user is pressing the snom dnd button.
yes, the phone is being provsioned by typing mac into pbxnsip.
Probably the usage of the snom 360 sidecar will also expose this problem (see the other topic from today). There must be something wrong with the DND indication.
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In the previous question about CO lines. i am wondering if i need to set it at all, regardless of the caller id. the reason why i set it was so i can dedicate the first line to incoming only and the rest for/incoming/outgoing.
IMHO the whole CO-line emulation is a little bit historic. Today they are virtual anyway, and in SIP there is not really any CO-line. In large PBX systems, there is also no CO-line monitoring (think about companies that may have 120 lines). The only point I see in CO-lines is a easy way of parking and picking up calls. Someone shouting in the office "Joe, Fred is on line 3"... An attended transfer would also do.
Since we spoke about led's, i set the snom sidecar to BLF for all extensions. When a user is using their phone it lights up and shuts off when the user is done, that's good. When the user puts their phone to DND, it lights up only for a few seconds then goes away. Any way to make the LED light up until the DND is off for the extensions?That does not sound like a feature to me... I believe I have to dust off one of these sidecars.
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What about the second call that comes in on the second line and ringing? the receptionist doesn't hear anything, she just see's it on screen. We want it to ring out loud.
It could be that the phone only sees the notification for the LED. It should be easy to see if that is the problem. Check from the phone's web interface if it receives a INVITE message. If not, then the phone somehow is not included in the list of devices that should ring. From there on we can dig deeper.
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We have such a HT in an office, and it works with T.38 (though we have to reboot it from time to time in order to keep receiving FAX).
The CS410 PSTN gateway does not support T.38; therefore you must use G.711 for the FAX. If the FAX is in the office (same LAN) that should be okay. I remember there are a couple of settings for the HT regarding FAX. Maybe double check and make sure it does not use T.38.
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OK I got this to work.
What was the trick? Something that others also might stumble into?
I just have one more question about the address book. When I press the directory button it gives me the first 32 numbers but that is it. Is there a way to make it fetch more of the list from the server? Thank you!You can "drill deeper" by entering the next search digit. Then the PBX will refine the search. Don't push the digits too fast, you always have first to wait for the next XML page to show up before your can request the next one.
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I have tested here and it work okay. After you press ##, it goes back to the previous menu. Try deleting the audio_en folder and then download the files again from http://www.pbxnsip.com/download/audio_en_30.zip.
For CS410, there are special files available, just next to the ZIP files in the software download section.
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Our recpetionist has a snom 360 phone and i enabled 4 lines on 4 buttons. All this connected to an audiocodes gateway. i enabled all 8 ports on the audiocdoes by entering the DID's, the first DID on port 1 is our main phone number and the rest are the DID's the telco gave us. When a call comes in, the 1st line button flashes, the phone rings and the correct phone number of the call comes through. Now when the second caller comes in on Line 2 at the same time the caller id is wrong (It shows the did number i put in the audiocodes), the button on line 2 is flashing and there is no ringing out loud. How do i get the caller id to show up correctly on the second, third, etc..call as well as it make it ring? This is for Canada BTW.
Of course the PBX can only display what it gets. AudioCodes gateways are pretty flexible, there must be a way to make the AC send the correct DID number. But I am not the AC expert. In the end the PBX just uses the information in the From- and To-header to display where the call comes from and where it goes to (the Request-URI is used for the routing, not for displaying). My suggestion is to keep an eye on the From/To header and try to change the AC setup.
i set it both 10 digits NANPA, and 11 digits NANPA and no difference. i also set it to p-asserted identity. this on one trunk.Yea, this should make no difference. This is more for outbound calls.
i also have 6 lines and set the CO Lines to line1:i line2 line3 line4 line5 line6. should i have set this for FXO? is it needed?The CO lines do not influence the display of the caller-ID.
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Is there a way to make the default playback behavior to include the message envelope?
No. We had that in the beginning, but most customers were complaining they don't want to hear it. People are so impatient today!
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When we try to use the dial by name if the system does not find the person it starts to go through each and every person in the phone book. Is there a way to have it say that ext is not found.
Those numbers that should be invisible (also for the dial by name) must be listed in the setting "Accounts that cannot be called". And also, you can tell the PBX how many digits it should collect before starting to search ("Start Search").
Also it does not say you need to enter the persons LAST name, it may be obvious to some but not to others.The PBX should be searching in both first and last name. IMHO the whole first name, last name topic is debatable. It would be better to have just the name (just one string). There are so many middle names, McXXX and Prof., Dr. and von zu.
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I have a few doctor office that all have multiple offices. I'll use 1 for an example for this
Medical Practice
Office1 w/ co1 co2 co3 co4 / Dial plan is office1
Office2 w/ co5 co6 co7 co8 / Dial plan is office2
Office3 w/ co9 co10 co11 co12 / Dial plan is office3
Huntgroups (HG)
HG1 w/ 9255131111
HG2 w/ 9255132222
HG3 w/ 9255133333
CO lines work fine for outbound calls but don't show incoming calls and when placed on hold the lines don't light up and flash. We have the trunk groups as SIP gateways as we couldn't get the registration to work on the nextone for some reason. Please let me know if you need more info. Thanks inadvanced for any advise.
When inbound calls don't stick to the CO lines as they should that smells like a problem with the identification of a registered trunk. There are some service providers that ignore SIP URI parameters (which is definitevely not RFC-compliant), and then then PBX has problems matching incoming calls to the right trunk. Check if there is a "line" parameter on incoming calls. And also the PBX talks a lot about the trunk matching in the log file if the log level is high enough!
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We've setup a test domain for prospects to use and give them a limit on the cost to us. I want to allow a a few people to call 2 different countries to test with family. I have setup the following in their dial plan. They have no problems with calling domestically just these.
011380* -> *
011972* -> *
I also tried
011380* -> left blank
011972* -> left blank
They should be able to call anywhere within these countries.
That looks okay... Try turning the logging on and see how the PBX processes the dial plan when the user dials a number. Maybe wrong assignment of the dial plan to the user or maybe the wrong trunk in the dial plan.
). Maybe that will help to present the buttons in a way that is easier to digest.
....... feature on proxy routes call back to pbxnsip if the call isn't answered......
transferring calls does not transfer CID
in Call Treatment
Posted
This is a general SIP problem. Are you using snom? There is a RFC for this, but most devices don't support it. snom has some bugs in this area, abut at least in principle it is supported.