I have been experiancing DTMF problems with Trixbox since the 21.2-2 release, apparently there is a known problem and a patch..
http://bugs.digium.com/view.php?id=13209
Has this ever been addressed, I am still having this problem and had to switch to inband.
Its a horrible problem. Outbound DTMF is useless.
I'm anxiously awaiting the patch to be integrated into the distro.
We don't have the option of switching to inband as that will break incoming DTMF. Seems I have a choice of one or the other.
I opened a service ticket with my voip provider and they pointed me to the same bug you found.
Nothing we can do now except wait.
Who is the Voip provider? Maybe you can beat the problem before the patch is applied
I have considered downloading the Asterisk source and implementing the patch, I'm not sure how this would effect the Trixbox installation but unless they addressed the issue in the new 22 version (that I can not update because I get a missing library error). I have little choice. The provider in my case is Bandwidth.com.
Broadvoice and VoicePulse.
But, I don't believe this has anything to do with them.
Bandwidth.com and inband here as well.
Can anyone tell from the SVN when this fix will be released? Which version of Asterisk?
The page says Status Resolved -> Closed. It says nothing about which version of Asterisk has the fix.
I've been eagerly updating asterisk.i386 each time a new version comes out, and I'm up to 1.4.22-3 as of this morning, and we STILL have no outbound DTMF.
It's been going on a month now.
It looked like the change was put in Dec. 9th so anything after that should have it, unfortunatley I've been unable to track down which one that would be as far as I know the change was not even in 22-5 so it would be in 22.1 for sure but who knows when they will get that in the package.. Whoever puts the ISO together could probably answer this in about two seconds if they gave a ^%^& but since it does not effect everyone we are just kind of hung out there. ok I'll give whoever can answer this $1.. yea thats what I'm talkin about.
Unfortunately, I don't have a test machine.
I should have messed with that over the long weekend.
With PBIAF you get the source so it is just a matter of applying the patch.. I'm at the point where I'm just going to download 22.1 compile it and go from there....
Compiling from source is very simple however you need to make sure you also compile the add ons package and Zaptel.
You also need to be careful if the release your compiling is DAHDI only.
In all fairness PBXIAF has a hold on Asterisk compiles for DAHDI testing currently.
Scott
aka "Skyking"
If I could just tell from the SVN what release the patch shows up in... That would be useful.
I can't understand why that information isn't there. I must be in the wrong place.
I have implemented rfc2833 between a Trixbox 2.6.1.3 with Asterisk 1.4.22.3 and a remote Spa3102. I have to say its horribly frustrating...if I dial my Ipkall # to hit the trixbox server in order for it to contact the SPA and then pass some digits, lets says 12345, the SPA reports seeing some crazy numbers like 1122233455...This bug has been identified and fixed here ( http://bugs.digium.com/view.php?id=13209 ). I tried to apply the patch as suggested but was unsuccessful...I guess trixbox customizations was failing it...It appears that Asterisk version from 1.4.19.1 to 1.4.22.3 are the faulty ones but 1.4.23 include the fix to address this Dtmf passing issues...I would really, really appreciate it if trixbox would make 22.3 available asap or at least provide a way to apply the patch from digium.
I meant to say make 1.4.23 available
Fonality does not customize Asterisk, you have to compile Asterisk to apply a patch so I am not sure what you did.
It would be nice if Fonality would build 1.4.23 for us (Andrew ????), However it is not difficult to download the package from Digium and compile from source.
Scott
aka "Skyking"
I have kernel 2.6.18-53.1.4.e15
I followed the directions at http://www.lowe-family.me.uk/trixbox/Asterisk_compile_on_Trixbox.... and I did a yum install kernel, and yum install kernel-devel.
This installed kernel 2.6.18-92.1.18.e15. So now I have 2 kernels.
Rebooting the server tried to boot into the .92 kernel, which results in a kernel panic and hang.
If I boot back into the .53 and my server is back up, but when I try to compile anything, it says I don't have the sources for kernel 53 installed.
I don't have an option for the .53 kernel-devel...
I am therefore at a dead end and don't know what to do next.
You don't need to compile the kernel. The kernel dev packages that are dependency of Asterisk are in the repository and can be installed with yum.
Scott
aka "Skyking"
That's what I would assume, but a 'yum list kernel-devel' returns only one item.
kernel-devel.i686 2.6.18-92.1.18.el5 installed
No other choices.
SkykingOH would you be kind to elaborate on : "However it is not difficult to download the package from Digium and compile from source."
Thanks!
Basildane,
I am not sure what state your machine is in since you tried to compile a new kernel.
What dependencies are not resolved when you try to ,ake amd make install?
Ous and Bsaildane -
This is the reference I use to build Asterisk from source. Works well for me.
Scott
aka "Skyking"
After 6 hours, I finally (i think) have everything compiled and running.
I haven't tested everything yet, but I have to take a break for a while.
The first crisis was, there is no more chan_zap.so. It compiled chan_dahdi.so.
I finally found a switch to add to asterisk.conf (dahdichanname = no) and my zap channels seem to be up now.
I had to find the proper kernel-devel on some obscure website somewhere. After renaming them, zaptel compiled successfully.
Still no idea why I cannot use kernel 92. I'm still on 53.
Thanks for keeping me going, Scott!
Oh, btw, my DTMF is WORKING NOW!
Asterisk version 1.4.23.
I can hardly believe it! Woo hoooo.
Not sure why you had so much trouble with the Kernel-dev sources. They are in the repository. You should have need able to yum install kernel-dev
Here is the file in yum.trixbox.org/centos/5/RPMS
kernel-devel-2.6.18-92.1.18.el5
It's the 92 Kernel that you need.
Good catch on the DAHDI switch. Soon we are all going to have to bite the bullet and learn DAHDI.
Fantastic news on DTMF issue.
Scott
aka "Skyking"
I did install the 92 kernel. Then my machine would not boot.
It threw hundreds of errors, missing volumes, drives not loading...
I removed it, reinstalled it, same problem.
I tried installing it from the trixbox repository, from yum, every way I could think of. Same problem each time. In Grub, it showed Centos ...92, and trixbox ...53. Only trixbox 53 would boot successfully.
If you have any suggestions on troubleshooting it, I'll try them.
When I try and update the kernel to .92, on boot it reports:
/dev failed: no such file or directory
setuproot: error mounting /proc
setuproot: error mounting /sys
Kernel panic - not syncing: attempted to kill init!
Then it hangs.
That is outside my abilities (and desires) to troubleshoot.
If I had these issues I would pay KodaK to do it for me.
Scott
aka "Skyking"
You may want to try different kernel boot parameters.
Try
acpi=off noapic
or a combination of the two.
Hi Guys,
Can I know if asterisk 1.4.23 will be available if I run yum update in trixbox?
It is not available in the repo, it's 1.4.22
However check your PM
Scott
aka "Skyking"
Anyone have an idea when 1.4.23 will be available in the repo? Also I noticed that 1.4.24 was just released with some more bug fixes, would it be better to upgrade to 1.4.24 if I am going to manually install the update?
I am having the dtmf issue with multiple tones being sent.
I am using g729 with bandwidth.com and have 2.6.2.2 with asterisk 1.4.22-3.
I was considering installing the patch for now, but I am not completely sure of the process to install the patch...
Thanks in advance
I upgraded to 1.4.24 last night because of the problem with "General Call Pickup" not working in 1.4.23.
There are instructions in the asterisk file on how to upgrade. Download the asterisk file, extract it and look at one of those readme files.
Keep in mind 1,4-24 needs DAHDI, Zaptel is not supported anymore.
The guide in the wiki for taking a 1.2 machine to 1.4 is a good tutorial for compiling Asterisk, Asterisk add-ons and DAHDI from source.
DAHDI is Zaptel renamed so those instructions flow also.
Why don't you load the system on a Virtual PC and try the procedure before moving to production?
Scott
aka "Skyking"
I decided to take the jump to PBIAF, one of the reasons was the ease of recompiling the code and the non fork of FreePBX which I feel a lot safer with. If you choose the U option when compiling the Asterisk you get 1.4.24. I did this on two systems also loading the 64bit versions and no problems.. I was able to set DTMF back to rfc2833. One of the two systems I moved over has a TDM400P card and Zaptel compiled and runs with no problems.. the hardest part is getting evrything over to PBIAF from Trixbox and I do miss endpoint manager but I'm hoping FreePBX gets something in that area.
I truly wonder why nobody from Fonality or the trixbox CE team is saying anything about this ?
adabbas--Most of us here at Fonality are really busy at the moment :) It's still normal business hours and the CE team is totally swamped! We have tons of support calls, lots of tickets to resolve, and many problems to solve! I'm not a trixbox dev, but I can tell you that implementing a patch to asterisk is something that must be tried on various systems and thoroughly tested before it is included in a routine update. Since asterisk is the base and the largest component of trixbox, any upgrades to it must be well-tested and tried before release. Luckily, we have the best engineers in the world, and I'm sure they are working on a solution :)
If you have the best engineers in the world, why are there issues with the trixbox system after being public for several years already, 2.6 being downloaded for almost a year already. Other projects like elastix and especially pbx in a flash release the latest version of asterisk immediately. Trixbox has been under 1.4.22 for quite a while. 1.4.23 has been out for quite some time and users still cannot update with yum.
This is what happens when two professionals, Kerry and engineer tim, are let go.
Sorry did not mean to sound ungrateful; the 1st thing I thought when I saw this post on 01/04/2009 was "soon enough there will be a fix from the trixbox team for it", now after about three months I wonder why nobody even bothered to say that “we are working on a fix for it”.
I just love trixbox so much and hate to see people jumping elsewhere because of those small things.
A couple of comments
This DTMF issue is a problem that anyone using SIP trunks is having, unless they have switched to inband which by doing that can create other issues...
As someone wrote above here, although I like trixbox, it seems that the other system providers are getting the asterisk patches or updates out when they come out. If it takes 6 mo's or a year for an update to a problem that many people are having then I will be moving to another system and abandoning trixbox! Spending days reading to try and find a solution for a problem that has been known about for 6 mo's is just not something that I am willing to do much longer.
I am working on a test system, and so it doesn't affect my production system.
I am going to attempt the 1.4.24 upgrade right now and hopefully everything goes well. If not if someone can post some updated step by step instructions now that things are with the new DAHDI that would be great! It would make a good wiki how to since there have been a lot of changes with asterisk and we are not getting updates from fonality. (I understand the number of engineers on CE is limited, but it seems that fonality is no longer providing CE updates.... )
It almost seems as if fonality waits for the people in this forum and others to find the solutions for them before they incorporate it into a release...
Up to 25% of our outbound calls, are a mess when we try to call other IVRs and press buttons on our phones.
zapata.conf
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
relaxdtmf=yes
echocancelwhenbridged=no
echotraining=yes
rxgain=15.0
txgain=-5.0
group=0
callgroup=1
pickupgroup=1
immediate=no
relaxdtmf and echo training per above, seem to help.
But we still have the problem.
Aastra phones, 53i on 2.5.1.200
Kernel Version 2.6.18-92.1.18.el5 (SMP)
TB 2.6.2-1
Asterisk 1.4.22-3
Sangome A200D w/echo cancel and fairly current drivers.
Dell Poweredge T100 dual something w/2GB ram and light load
What seems to help: As soon as the other IVR picks up, you start talking for a few seconds. Then press buttons, and it is clear. But not every time. Sure seems like an echo cancel / training problem.... but this post makes me wonder if we need to patch asterisk.
Randall @ fontality has been great help... but we are searching on this one.
Patching asterisk appears most tricky.
We only make outbound calls on POTS lines for now.
One test is maybe setup for SIP outbound and try calls....
thanks for ideas
bill
I am trying this setting :
from Sangoma wiki
Edit your /etc/wanpipe/wanpipe*.conf file and change "TDMV_HW_DTMF = YES" to "TDMV_HW_DTMF = NO" and then stop asterisk and run "Wanrouter restart" and then start asterisk back up and see if this resolves the issue.
Member Since:
2007-01-26