Vodia PBX
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I don't believe that is a concern, it is more the functionality without having to double up on the number of extensions licensed to allow people to have this ability.
Well he can go to his own mailbox, push 5 for recording a message and then move it into his own mailbox. Less than ten key strokes
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We have a CS410 running version 3.1, customer called several times complaining that many times approx. 1 minute in to the call the Snom 360 phone displays timeout occured and the call gets disconnected. We are using the built in gateway.
The call log shows the call however the duration of these calls are blank.
Try turning off polarity reverse detection. That is causing such effects. I guess technically the call is never connected, so the phone disconnects the call.
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I have a client that is looking for a simple dictation option where they would record a dictation and have an email sent to them. Is there a way to do this (* code or option in the voicemail menu) without setting up a separate extension to call into?
How many buttons does he want to push?
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Is there a work around for this problem that we can use for 3.0.1 until 3.1 is finalized?
You can drop the attached file into the html folder. Don't forget to remove it when finally upgrading to 3.1.1 (or later).
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Hi Support
I've seen this one before on the forum and I'm sorry to tell but I have the same problem .
No more Licenses available when I try to create an extension or an account.
System Info;
Version: 3.0.1.3023 (Win32)
License Status: pbxnsip Europe - S-Point
License Duration: Permanent
Additional license information: Extensions: 2/4 Accounts: 5/13
I received a new license today and pasted it in to the license field
Whow thats strange. Do you have too many domains? Trunks? Having the 3-minute key is nice, but when you start using a real license key it sometimes comes asa bad surprise that there are already too many resources for a regular license key.
Do you see anything like this in the log? "License suspended: There are too many domains"
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Advice please: how to turn off "Rewrite global numbers" function? We don't use +/0/00 at all, but the system add "+" before ANI, and trying to use 0 and 00 prefixes on DNIS.
Version: 3.1.1.3091 (Win32)
Clear the country code and it will be shut off. That whole global number topic was a bigger cleanup round than we thought, but we are seeing light at the end of the tunnel now.
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Thanks a lot for the information about STUN. Nevertheless I'm running my system (as I maybe decribed in my post above) with a keep-alive time of 30 seconds. At the beginning it works really good and renew the registration after every 30 seconds. When I dial my sipgate number from my hardware phone to connect to my MS Speech Server application (yeah, the routing works now fine), the registration get lost after this call and only a reboot of my system with in general a new IP from my router (I used DHCP locally) will help. Anyway, that can't be the solution. When I have a look at the output from Wireshark it seams that the DNS of my dsl modem can not resolve the connection to sipgate, but that happends also before, or?
The PBX send REGISTER to sipgate, but sipgate do not answer with 401 that the PBX can send REGISTER with the authorisation (in my opinion from the analysis of the Wireshark output).
If it will help I can upload the log file of Wireshark.
I think it is a little bit strange.
Cheers zazi
Is your DSL router doing DNS? There are many DSL routers that are not able to deal with non-DNS A requests (SRV, NAPTR, AAAA).
In general, if you have to get a new IP address then I would really really recommend trying another router. There is just so much buggy stuff in the low-end router market that you can burn a lot of time pinpointing the problem just to find out that the NAT implementation was just a little bit toooo pragmatic.
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why is audio files in two directories? /library/pbxnsip and /private/var/run/pbx? I copy now the new audio files to second directory and it works!
Yea, the Mac installation is a little bit screwed up. We need to focus on a new Mac installer.
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.. and why are't new audio prompts attached to installation of pbxnsip for mac?
I must download separated from install (but I did not know this, until today): http://pbxnsip.com/download/audio_en_30.zip
:blush: We are just not big experts on Mac installations... In a perfect world, the Mac installation would just run on a shell script; but we are not sure if that is an option.
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Yeah, I checked out again my other test PBX (3CX Phone System). There the registration will be refreshed after my configurated period of time successfully. So maybe it has something to with my configuration at PBXNSIP, or I don't know. 3CX make use of the STUN server of Sipgate. Unfortunatelly, this isn't any more available at PBXNSIP. The big disadvantage of 3CX is that it does not work together with MS Speech Server 2007. PBXNSIP does it, fortunatelly.
Yes, in the beginning we also offered STUN support for clients behind NAT. There was a hope five years ago that STUN would solve the problem overcoming the routing problems with SIP. But we just drowned in support explaining customers what the difference between full-cone NAT and symmetrical NAT is and asking customers to try new routers. That's it! STUN is not the solution.
Today it looks like SBC will will win the race and IPv6, that's the "inconvenient truth", will be the only way to have true peer-to-peer SIP media. Also outbound looks promising (see http://tools.ietf.org/html/draft-ietf-sip-outbound), this will solve a lot more problems that the 2003 STUN did. Of course, we put everthing into the pbxnsip PBX already
.I don't believe that a free, home-compiled SIP proxy and some good marketing is enough to build the ITSP money printing machine. The truth is that a lot more is required and carriers do have to spend money for SIP equipment. Time will tell, and customers are making their vote on good and bad service. Also, a big question mark is QoS. Some say that a solid carrier need to have control over it, otherwise voice hickups when emails are being uploaded are only a question of time. At times when 100 MBit/s are available for a few bucks, there is no easy answer for this.
Keepalive Time: 600I would try setting that to 30.
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What you can try is to set the "Keepalive Time" explicitly to something like 30 seconds and see if that makes a difference. 600 seconds is too long for NAT unless sipgate sends some keep-alive traffic. But not all firewalls treat keep-alive traffic from the outside as valid keep-alive, maybe you are the lucky one with such a firewall.
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All WAV files are lowercase and also have permissions to read. grr.
what can be the problem?
Check if the PBX runs in the expected directory. See the status screen of the PBX.
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Looks bad...
If there is absolutely nothing coming back then maybe your firewall blocks the traffic? Even if sipgate does not offer SBC service (which is neccessary to run the PBX behind all kinds of NAT unless you have a routable IPv4 or IPv6 address, see http://en.wikipedia.org/wiki/Session_Border_Controller), their edge proxy should respond with something.
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Logfile
[1] 2008/11/23 06:28:38: UDP: TOS could not be set
[1] 2008/11/23 06:28:40: Last message repeated 4 times
[5] 2008/11/23 06:28:40: Tuning to new SSRC
[5] 2008/11/23 06:28:43: Last message repeated 2 times
[5] 2008/11/23 06:28:43: INVITE Response: Terminate 36f2255e@pbx
[5] 2008/11/23 06:28:43: Tuning to new SSRC
[3] 2008/11/23 06:28:47: Could not open WAV file audio_en/aa_receive_callback.wav
[3] 2008/11/23 06:28:47: Could not open WAV file audio_en/aa_leave_message.wav
why could not open WAV file? both files are in /Library/pbxnsip/audio_en/ (Mac OS X) and what is TOS?
The TOS is not so serious and is fixed in newer versions. The WAV problem may be because of upper/lowercase problems? Check if there is anything uppercase in the file system. And check if the PBX process has the permissions to read.
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I have a doctor's office that is having a very annoying problem with Agent Groups that can be reproduced every time. A call comes in from the PSTN into a ring group (the front desk area). Someone answers the phone and then they want to transfer the call to the Agent Group for the optical department (3 phones in the group). Two problems happen:
1. The customer hears dead air instead of on hold music when the call is transferred until someone in the optical department answers.
2. Once the call is answered by the optical department the other 2 phones continue to ring until someone picks them up (they hear dead air) and hangs them back up.
This happens 100% of the time and can be easily reproduced. This can also be reproduced by having a ring group that's final stage is to go to an Agent Group however the caller hears the on hold music like they should.
If you call the Agent Group directly from any extension directly it works correctly. It has something to do with being in a hunt group first from what I can tell.
What kind of transfer is this? I guess attended transfer? The blind transfer should work and IMHO is a good workaround (because the agent group will always take care of the call).
The attended transfer is very tricky as we have to transfer the media state as well. We already had problems with conference and regular calls (which should be solved), and we now need to look into agent groups as well.
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you know how if you call from a user to another user it says "the person is unavailable to press 1 to leave a message or 2 for a callback" or something like that.... how can i disable this?
There is a setting in the domain called "Offer Camp On". Turn it off and the prompt will disappear.
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Does anyone know whether pbxnsip uses RFC 3856 to know the state of phones which are connected to it or does it use some other way to know phone presence?
The PBX is at this point just a presence agent that receives the presence information from a user-agent and then notifies it to subscribed parties. It does not look at the presence document itself.
A UA that wants to publish its presence state must use PUBLISH and the event must be set to "presence". Pretty SIMPLE!
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Always make a backup before removing something...
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I attached the log in the first post. No idea what most of it means, I was hoping it showed you what you needed.
Well, that log did not reveal what the problem was... It would be good to see the SIP packets. See http://wiki.pbxnsip.com/index.php/Log_Setup on how to turn that on.
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Agreed - it still dialled ok though! I think the @ symbols was my fault, I had it setup to add @pbxnsip.wm.local when it dialled, and it was doing it each time I tried to resend.
And what does it say now?
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INVITE sip:xxxx@pbxnsip.wm.local@pbxnsip.wm.local@pbxnsip.wm.local SIP/2.0
That looks strange. Too many @ symbols... And it is not a valid SIP URI.
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Debian revision? Also, please address the logging issue? Lastly, I still have not had an answer on forcing outgoing ANI changes as currently it sends the incoming ANI out for redirected calls and that is always not possible as some vendors make your ANI be one of the known DIDs?
Debian: http://pbxnsip.com/protect/pbxctrl-debian4.0-3.1.1.3094
ANI: Did you try choosing the "Remote Party/Privacy Indication"? Maybe just set it to "No Indication"?
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Using auto provisioning with the snom 320,360 and 370, we have good success. However, when we try to auto provision snom300 we do not get the buttons assigned, In the generated directory, the file "snom_300_fkeys.xml" has a length of '0', so naturally it won't work. Any suggestions why the system is not generating this file.
We are seeing this on all systems we have or support.
Yea, that problem is fixed in 3.1 - but we are still working on getting a good 3.1.1 that we can release publically.
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Yes I realize that, it then takes all the paramters from this file. But I loose the unique items per gateway that I require, such as the account, name etc that I get when the system uses the mac address to identify the spa2102 from the generated file. Is it possible to get the unique parameters from the generated file and the common from the html directory, I have not been able to do that, with with the spa 2102 or the snom phones.
Sure you should be able to do that. Check out the above file, there is for example {account} in the file that depends on the MAC.
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Well, you can lock the phone, just like you lock your cell phone.