RE: max channel PER TRUNK vs. PER PROVIDER 11.10.2008 00:38

Forum: Providers

Apologize me for that oversight.

Anyway, my thinking is the same of Dgerber: I found illogic put that settings in the trunk section.

Some question:

If I have more trunks on the same provider, and I set different values in the "Maximum Channels" of every trunk, which value will be considered ?

What is the reason why you can't enable that setting on the trunk ?

It will be very useful.

Thanks for your support

I don't want to pay support for your bugs! 10.10.2008 11:10

Forum: Providers

I am very frustrated of your service.
Ev'ry time I change a configuration nothing works correctly.

FIRST BUG:
Trunk1 with max one channel enabled (in/out)
Trunk2 with max one channel enabled (in/out)

Group#1 with two extensions (example: 100-110) in ringall mode.

Inbound route destination for trunk1 : Group#1
Inbound route destination for trunk2 : Group#1

Incoming call on trunk1: the two extensions ring together, extension #100 pick up the call.
During the conversation on trunk1, I tried to call the other trunk (trunk2) and the extension #110 don't ring and the caller (me) don't hear nothing.

This is the log of my provider:

-- Got SIP response 480 "Temporarily Unavailable (Call limit)" back from 88.198.18.239
Transmitting (NAT) to 88.198.18.239:27571:
ACK sip:01234567869@88.198.18.239:27571 SIP/2.0
Via: SIP/2.0/UDP MyProviderAddressIP:5060;branch=z9hG4bK1d179154;rport
From: "mynumber" <sip:mynumber@MyProviderAddressIP>;tag=as048971fc
To: <sip:0123456789@88.198.18.239:27571>;tag=as555cc640
Contact: <sip:mynumber@MyProviderAddressIP1>
Call-ID: 35aa2db64a877a4b4e9ec8844f53471a@213.204.0.81
CSeq: 102 ACK
User-Agent: phone
Max-Forwards: 70
Content-Length: 0

Question: why your service think that the call limit on that trunk was reached ???

So I tried a workaround, because I need to work today with my phone number, and changed configuration in this way:

SECOND BUG:
Trunk1 with max one channel enabled (in/out)
Trunk2 with max one channel enabled (in/out)

Inbound route destination for trunk1 : ext. #100
Inbound route destination for trunk2 : ext #110

Calling one of the trunks, the operator's phone ring but the caller don't hear nothing! If operator answer the call, the comunication is estabilished regularly.

Have you enough, or I need to continue with your nice bugs ?

I hope in your fast support.
Thank you

Issue with "Not available" message recording. 09.10.2008 18:08

Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

I enabled the voicemail in a new extension and I recorded the message through the ip phone (Linksys SPA922) like usual (*97, 0, 4 etc.etc.).

Finished the recording, I check the message and confirm the save request. All seem ok !

But, calling the extension, I don't hear my message but only 1 or 2 seconds of silence followed by the "beep" tone that advise the starting of the voicemail recording.

The message is recorded regularly. This mean that the voice mail service work fine.

But why I'm not able to record my custom message ?!!?!?

Thanks for support

RE: Ghost system recordings ?? 13.08.2008 17:47

Forum: Bugs

Now all work fine !

Thanks for support.

Ghost system recordings ?? 12.08.2008 02:23

Forum: Bugs

I uploaded a new system recording then I set a new digital receptionist menu for use it.

Testing the functionality, I don't hear the correct recording I choose in the receptionist setup.

So, I deleted all recordings leaving only the one I need but.... I still hear the old recordings.

Maybe is there any ghost recording in your system ?

Could you check what's happen ?

Thanks

RE: No more logging with extension account ?!? 07.08.2008 10:00

Forum: Miscellaneous

I discover where is the problem: i used the extension password instead of the voicemail password.

:-(

Sorry for inconvenience.

No more logging with extension account ?!? 06.08.2008 10:07

Forum: Miscellaneous

I'm not more able to log into web interface using my extension account.

The system notify the following message: "Incorrect Username/Password".

I checked the parameters and all is ok.

Could you help me ?

Thanks

RE: All my extensions are down !! URGENT !!! 22.05.2008 00:32

Forum: Bugs

Zitat:

Originally posted by i-p-tel Motivated by the last two outages on May 1st (www5) and May 19th (www3) we have installed a state-of-the-art monitoring system that can also detect failures in SIP registration. You can obtain status info at http://pbxes.org/status.

These are the informations that help us to trust in your service.

In my honest opinion, I think that support is the only one thing that make the difference !

I hope you will improve your service and support in the future.

Best regards

19.05.2008 19:34

Forum: Bugs

Zitat:

Originally posted by archmei I have the same problem also with this account, my ip phone not register. Please resolv this problem soon!! Tanks!

Try to change the server in the Personal Settings.

Good Luck....

Thank you so much for support.... 19.05.2008 19:33

Forum: Bugs

Maybe the support today was on the beach.
I hope that sun was hot and water very refreshing.

Instead, in my office, it was storming.

All my voip-users, strongly desire my head !

I understand that could be some issues in service, but a condition that I couldn't accept is this incredible silence from the support (supposed that there is a support!).

A serious company, should have apologized his customers for issues, communicating that all resources are hire to solve quickly.

Maybe, I have an incorrect vision of a serious company.

Anyway, I wish to thank you all forum people that suggest a workaround.

Have a nice day... if you could.....

All my extensions are down !! URGENT !!! 19.05.2008 12:23

Forum: Bugs

From this morning, all my extension are down.

The registration fall down in erro 408 (Request time out).

Please check my problem because I had four offices that can't work.

Thanks

Call monitor... how to ? 06.05.2008 00:13

Forum: Miscellaneous

Two question:

1)
How could I purge all the calls in the monitor section ?

2)
How could I filter the results in the call monitor ?

For example....

If i want to see only the call where destination is the group #2 , what I need to write in the search box ?

Thanks for the answer.... (i hope).

Understanding System Log. 29.01.2008 01:35

Forum: Miscellaneous

Is there any reference guide to understand the new system log feature ?

Thanks

RE: Technical answers regarding vocoder selection & Re-Invites 29.12.2007 16:25

Forum: Miscellaneous

Zitat:

Originally posted by Diafora Caveat The above answers, cover the technical aspect of vocoder selection and the use of Re-Invites. They do not however cover how the "Usage" statistics are calculated, or anything else related to it. This should be treated as a general explanation of how vocoder selection works and how Re-Invites work in the VoIP universe.

Thanks Diafora for your explanation.

The only thing I don't understand is why you didn't answer my question #2, that was mainly the reason of my post.

Maybe is there particulary reason I don't know ???

I'm your customer, and I think there is nothing strange for me and all your customers ask to understand which parameters impact on the bandwidth usage to optimize the resources.

For example....:

assuming one hour of conversation between two SIP phones using G729 codec, estimating the well known 16Kbps usage, is it correct to state that your system will calculate an average bandwidth consumption equal to : 16Kbps x 3600 seconds ?

Moreover......

if I set the G729 codec passtrough enable to yes, is it also correct that bandwidth usage I above mentioned, will be strongly reduced ?

Thanks for your answers.
Best regards.

Clarification on bandwidth usage... 28.12.2007 17:44

Forum: Miscellaneous

I have some questions for better understanding the bandwidth usage and relationship with PBXes.

Question #1:

Who decide which codec use in a conversation ?

In other words.... if i set up my ip phone (or ata) to use a specific code (for example G711), it's certain that it will be used ?

Question #2:
What append when PBXEs receive a calling and connect the two points ?
The audio cross trough PBXEs (increasing the bandwidth usage) or it limits only to connect the caller and the receiver directly ?

Question #3:
Could PBXes modify the codec used before starting a conversation ?

Example:
The caller use G711 codec,
The called support two codecs : G729 and G711
In this situation, which codec will be used for comunication ? G711 ?

Could PBXEs force the G729 codec to reduce bandwidth usage ?

Thanks to all for support

Happy New Year !

24.12.2007 16:14

Forum: Miscellaneous

I seem to understand that you don't have any email address to communicate directly whit your company ?

Very strange for a communication company...

Anyway....

I need invoice for my premium account purchase for tax deduction.

I hope this could be a common request.

24.12.2007 03:47

Forum: Feature Requests

I'm interested too in an export call log facility.

With this feature, will be possible do statistics and call analysis.

I don't understand why there are so many problems to implement it.

I need email address for billing information ! 24.12.2007 03:20

Forum: Miscellaneous

Is there a particulary reason why in "Contact" section there is not a corporate address ???

I have some administration question.

How could I reach somebody in your nice corporation ??

Thanks

No BUSY TONE !!! Why ?? 07.12.2007 15:51

Forum: Miscellaneous

I configured a sip trunk with only one channel maximum.

I created an inbound route for that trunk.

If I call that trunk when another conversation is in progress, I don't hear the busy tone.

I hear only silence and, after some seconds, the call fails.

Could you help me to solve the issue ?

Thanks

29.11.2007 10:51

Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

I didn't know this feature. Thanks a lot !

And what about the call monitor ?
For how many time it store the calls list ?
Is it possible to eliminate all the calls list ?

Where I can find a complete guide to functionalities of Pbxes ?

Thanks