Vodia PBX
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But it would be nice in a multi-location site to be able to have 1 PAC console that could see the whole company.
You can start two PAC ...
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yea but if i want to sell snom G722 phones it has to convert to G711 as it hits the pstn...
so your saying this scenario is not CPU intense?
No not CPU intense.
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I have an issue on queue processing for PbxNSip;
This is when I have simultaneous callers on queue, and 1 caller is picked-up, the 2nd caller continuous to be on-queue instead of ringing the free Extension.
But when the 1st Caller hangs-up (or the Extension handling that call hangs-up), the call from 2nd Caller immediately went thru both Extensions.
Is there a parameter setting to make simultaneous callers on queue be handled by different Extensions? That is when the 1st Caller on queue was answered, the 2nd Caller will be sent to the free Extension and be answered as well.
That sounds more like a bug to me: Two guys in the queue, one of then ringing agents. That call is being picked up by *87, and the other call does not advance in the queue and just stays there in the queue listening to music. Right?
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How can i setup (if at all) that when a call comes in to the receptionist and she transfers the call to an internal user that the caller id should show the callers number and not the receptionist extension number?
this is an urgent issue as the person does not always want to speak to the caller however they think the call is coming from the receptionist!
Attended transfer or blind transfer? I guess it must be blind?
There is a RFC for updating the caller-ID. But the support in existing SIP devices is "weak"....
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HI, i have clients who are complaining that when people leave them voice messages it sounds very quite and difficult to hear...should i tell them to get a hearing aid or is there something i can do about it??
If you are using a FXO interface check the gain settings. Probably the signal is just too low. It becomes very obvious when you hear the loud IVR prompts and then a relatively low volume recording.
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i have clients complaining that when they make outbound calls many times they are getting an error beep beep beep after 5 or six rings instead of getting voicemail or something....it sounds like the call is timing out or something....
Any insight from the SIP logging? Maybe a request timeout after about 30 seconds?
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i am using a PRI line with a Sangoma PRI card and i changed the media to Message 180 but they still do not get ringback before the hitting the auto attendant.
pls advise
Hmm. Sounds almost like a problem with the Sangoma setup. Maybe a WAV file missing there? I know a couple of Sangoma setups that work without problems.
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Cant Wait, Thanks.
If you like give it a try with http://pbxnsip.com/protect/pbxctrl-3.1.1.3089.exe (Win32).
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The codec that is being negotiated is G.711. Do I need to change the codec?
Well, that is the codec for sending the media (which is fine, don't change it). What is the codec for sending RFC4733? Most of the time it is something like 101/telephone-event.
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I try to monitor extensions in a different domain as the default and tried to put the username in as username@domain but the PAC translates this as username@domain@ip address of server.
Any way to do this?
I downloaded the latest.zip so that should not be the issue.
Thinking about hosted environments that sounds like a feature to me... You should not be able to spy on your neighbors!
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I saw that provisioning account... what is its purpose?
Make it possible to provide a admin password for all phones in the domain.
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yea but if i want to sell snom G722 phones it has to convert to G711 as it hits the pstn...
so your saying this scenario is not CPU intense?
Well that's my point: G.722 and PSTN don't make sense. Then better use G.711, because PSTN is G.711.
Don't worry about CPU. I would say G.722 is at least ten times faster than G.729A.
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Unfortunately without the NAT support the SIP trunk will register and be able to recieve incomming calls but then you can't dial out.
As far as I know Mytel still belongs to the service providers who do not provide any kind of session border controller (SBC) functionality to their customers and instead requires that customers have a public IP address. Trying to workaround this fact will not make the solution more stable... If they require a public IP address maybe you should listen to them and get one for the PBX.
Why the SonicWall has only one registration is just another mystery. I guess we can spend a lot of time trying to figure out what happens between the service provider and the firewall. I could imagine it is something stupid like the service provider associates a registration with a IP address/port and that is why you can have only one registration.
A packet log from the firewall on the public interface will make this problem visible. If you have the time to troubleshoot this, put a Ethernet hub on the public Ethernet interface of the firewall and record the traffic with Wireshark. Then we can all take a look at this and find out if the problem is on the service provider side, the firewall side or somewhere else.
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It would be nice if we could have a service login. which would basically give domain admin rights only.
I dont really like the idea of my techs having the admin login, but i dont want to burn an extension per domain.
and it would be nice if on a domain by domain basis we could disable processor intense applications if we wanted. such as recording, meetme etc
That with the admin login will be difficult, just because we need somehow an ID. Maybe the provisioning account in 3.1 in the domain would be a candidate for this.
The CPU intense application is a valid point. But this cannot be a binary decision. We need to say how many calls can be recorded per domain and how many people can be in a conference per domain. A call in the "maximum calls" needs to be weighted according to the complexity.
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How many of these calls would you date put on 1 box?
We are thinking of offering HD audio to our fiber customers. Not sure how intense the trans coding will be.
G.722 is pretty easy. Transcoding is not an issue of CPU performance.
G.722 is not G.722.1. G.722.1 is a Polycom codec; that codec has a similar complexity like G.729A.
The bigger problem is that transcoding reduces audio quality. Remember that both codecs represent the audio in 64 kbit/s and that formatting the audio from one representation into another reduces the information. Therefore, the PBX has a strong interest in trying to avoid the transcoding anyway - that's why the PBX tries to UPDATE or Re-INVITE after an attended transfer; the typical case for running into a G.722 transcoding situation.
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What info do you want to see?
It would be interesting to see what DTMF codec has been negotiated and if the DTMF is inband what level and noise the DTMF has.
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all my phones are plug and play! i have approx 80 registrations!
Hmm. It could be the we have to provison this specific device a different firmware than the other phones. I remember something like the conference phones need a newer firmware than the good old 2.2.2. If that's the case we are a little bit in trouble, because we have only one tftp directory.
Hmm. Maybe we need a kind of "virtual" tftp directory for specifc devices. Otherwise I don't see how we can have different firmware versions for different devices! Ouch.
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I am using the same version, on the same platform...and am experiencing the same issue.
I ticked off "yes" to all the record options (I'm assuming this is how you record all calls), yet when a call comes in from PSTN and is redirected back out to the PSTN...There is no recording.
Should the PBX record calls in this fashion, or am I missing something?
Well, it is a little bit debatable if that call that should be recorded or not. But probably it should - as the caller probably ideally has no idea the call was redirected. We need to put that into the code. 3.1.1 will have it.
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So using your example the command would be:
http://pbx/reg_status.htm?save=save&send_recording=true
Is this correct?
The question remains why is the variable provided in the Linux pbx.xml and not the Windows pbx.xml of the same version? And how does it become "visable" or will the above command make it appear in the file?
Thanks!
Yes that looks correct. There should be no difference between Linux and Windows; the differnce is probably just when you saved something the last time. After using the link above, the file should contain the setting that you want.
We are piling more and more "hidden" settings up. Usually we don't want that customers play with these setting (because it makes support more difficult); however it is always good to be able to make a change when neccessary without changing the code. That settings above would be a candidate for becoming visible...
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i have a lot of traffic on my server but this is it i think...
[6] 2008/11/13 13:07:26: PnP: Receive 0004f21dd267-app.log:
1113120323|so |4|02|[soNcasC]: SoNcasFailOverPending - DoItNow !!
1113120323|sip |4|02|SipStartFailOver 1
1113120402|sip |4|02|SipOnEvNewWorkingServer User 1, old 0, new 0, expire 0
Okay, so the phone does go to the PBX...
Did you drop the Polycom files into the tftp directory? And also, is there at least one extension set up for plug and play? Check the "generated" directory if you can spot the phone's MAC address there.
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hmm...not too familiar with Linux but ill give it a shot...but if possible please add the functioin of adding wav files to agent groups in furter edition of the software.
You can also use psftp on Windows (which it putty), no need to learn shell scripts in Linux!
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SIP Trunk Registration Problems when running more than 1 trunk.
I having problems trying to get two SIP trunks to register together and stay registered.
Setup Info
Gateway Sonicwall NSA 2400.
Nat Enabled YES
Operating System: RedHat Enterprise Server RHEL5
PBXnSIP Version: 3.0.0.2998
Licence status Permanent
ISP: TPG
ITSP: Mytel
License Details
We purchased the office25 pack from Alloy Computer Products
It's supposed to support 2 Trunks by default plus we have purchased an additional trunk because we intend to run 3. We have installed the new license over the top of the old one as instructed.
Problem Scenario
When running just 1 trunk the system seems reasonably stable.
When I configure the additional trunk (Same ITSP) the 2nd trunk registers but and after a few seconds then the first trunk drops it's registration and will NOT re-register.
So far I have tried:
1) The usual rebooting procedure but it doesn't help.
2) Re-Installed the system from scratch.
3) Other ITSPs IE: Engin, Comvergence and ISPhone
4) Checked with the ITSP. There are no restrictions on multiple SIP registrations from the same Public IP.
5) Various routers without firewall enabled
6) Configured a second LAN adapter with a Public address
and I STILL get the same results.
Please can anybody help ???
I would disable the NAT support on the firewall to see if that is the problem.
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it is setup to alway allow.
So do you see any traces in the log that the phone tries to get to the PBX? Also check http://wiki.pbxnsip.com/index.php/Polycom...
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do i change this from the phone or from the pbxnsip server? all the phones are connected through PnP. also are there any disadvantages to switching to the TCP protocol?
There is something at http://wiki.pbxnsip.com/index.php/Polycom#...transport_layer, maybe just give that a try.
Voicemail Tinny
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I would double check the setup of the card.