Conference Bridge Broken

[Sippy]

At some point in time during the last few weeks my conference bridge has stopped working. When I dial the number the prompt works. It says "please enter the conference pin number". When you enter the pin number it says "That is not a valid conference number please try again"

I don't know how to pull up log numbers, but I did update the scripts and update fixes and it is still not working. Should I update the source also? I was reluctant to because it is a live system.

 

[jmullinix]

I have had a lot of trouble with in-call DTMF in the last couple of weeks. Try changing the DTMF setup on your box and extensions. If the conference login doesn't get proper DTMF, the login will fail.

I don't have trouble with dialing the call. The failure occurs with in-call DTMF.

 

[Sippy]

what setting for dtmf

rcf2833 is what my extensions are set to. Is there another setting I need? Thanks for your reply

 

[jmullinix]

Try inband or info, depending on your phone.

Turn off the login on the conference and see if you can use the bridge. If you can, that will isolate your problem to DTMF issues.

 

[Sippy]

No Joy Yet

I setup a conference without any pin numbers and I am getting the same message "That is not a valid Conference number please try again"

Also changed dtmf to info and inband with the same results.

 

[jroper]

Hi

This sounds like a timing source issue.

when you do zap show channels, so you see a channel called pseudo, or do yo have any zap hardware in the system?

Joe

 

[Sippy]

Thanks

here is the output

Chan Extension Context Language MOH Interpret
pseudo from-zaptel en default

I do not have any hardware in my system. I have sip trunks from Vitelity and from VOIP.MS. A few weeks ago the conference bridge was working fine. Should I update the cent os. I have only done the updates in freepbx plus update scripts and update fixes. Also it doesn't matter if I call in from outside or an internal extension.

Again thanks for the help.

 

[Sippy]

Still no Joy

How can I troubleshoot this? If you have an idea I would like to try it. If I update the cent os will things break?

 

[Linetux]

You can't do conferences without having a timing source - you'll get unpredictable results (which is what you're seeing).

I use one of Sangoma's USB devices to provide timing for my main PBX that doesn't have any PSTN interfaces in it:

http://www.sangoma.com/products/hardware_products/specialty_tools.html

That's been working great ever since I installed it. If you want reliable conferences, you either need a timing source with Asterisk, or you need to move over to FreeSWITCH which doesn't have such requirements.

 

[Sippy]

Fixed

After reading a few other threads, the command

service zaptel start fixed the problem. How can I make sure this starts on every reboot?

Nevermind I used Webmin to run the service on startup. That fixed it

Thanks for all your help.

It takes a village to keep the monkeys fed.

 

[Linetux]

Just FYI, you haven't fixed the problem. If your end users don't expect reliability, that's fine - but if you're trying to build a reliable system, you have to do it right.

 

[Sippy]

zapteldummy

Is the zaptel dummy not reliable? My system worked fine until that got turned off. In Webmin that service was not setup to run on startup. It uses the USB module as a timer source.

I am curious as to why that wouldn't work. We use a t-1 that uses an outside source for clocking. (Stratum clock)

You are causing me to have FUD.

 

[Linetux]

Where is the T1 coming into the picture here? The T1 is a perfect timing source. So is any analog card.

No, ztdummy is not a good enough timing source for conferences.

 

[Sippy]

Where is the T1 coming into the picture here? The T1 is a perfect timing source. So is any analog card.

No, ztdummy is not a good enough timing source for conferences.

Thanks for your input. Why is the ztdummy unreliable. If it works like mine does it should work. I have had this system for 1 year and it always worked until I did an update a few weeks ago. After I figured out the Zaptel was not running with yours and other input on this forum, I simply restarted the Zaptel and everything is working fine. Could it be related to the type of hardware in the PC? USB devices have been erratic since version 2.

What makes the USB device that you referred to reliable? I do not want to stick a device in a USB port and have someone come along and remove it.

 

[jroper]

I suppose it depends on your experience.

For some, ztdummy is fine, for others, not. It depends on your application, and the underlying hardware.

If you are happy with it, then leave it be.

Joe

 

[completetech]

I am wondering why a USB timing source is needed??? can someone explain it a little more and also possibly which one to use and how to configure?

 

[Linetux]

The USB module is used if you don't have another hardware source (PSTN card, PRI card, etc).

The problem is that without an accurate timing source, things get out of sync. You might not realize it, but during a meetme conference, the zaptel module is probing the timing source 1000 times a second.

Then there's this nasty little issue about RTC (Real Time Clocks) being based on ^2, which means you get 1024Hz ... not 1000 ... which might be okay sometimes, and not others.

This is an example of x86 not able to work well with telephony - and why cards (and other devices) have specific chips designed to work at specific clock speeds - to match what needs to be matched.

If you have no zaptel hardware, you're dealing with 20ms frames. You're getting a small amount of jitter caused by a 1024 -> 1000Hz conversion - it's USUALLY not significant, but SOMETIMES it comes and bites you in the butt. And when it happens, you look bad.

I'm oversimplifying some of this, but I hope you get the idea.

So that's why I always use a hardware timing source if it's something that I want to have faith in - in other words, something that's put into production.

 

[jroper]

Hi

As an addendum to what Linetux said, which I agree with, basically, it comes down to a cost benefit analysis depending on the use of conferencing.

However, on later Kernels than those supplied with CentOS, e.g. those found on Debian, Asterisk uses the realtime clock for a clock source, rather than the USB system.

Additionally, it may be worth referencing this thread for further information as to how things will improve,

http://www.russellbryant.net/blog/2008/06/16/asterisk-16-now-with-a-new-timing-api/

Joe

 

[completetech]

thanks for the explanation. I got a little more grasp on it. but from he article Joe posted a link to. I should be able to get away with now not needing one with asterisk 1.6..so I see. but then again $65 for a Sangoma USB voice sync is not to bad to be on the safe side.