[asterisk-users] DTMF Problem
bilal ghayyad
bilmar_gh at yahoo.com
Thu Oct 2 12:38:30 CDT 2008
Hi;
This problem I suffered from it for long time, it needs some work from ur side to resolve it, I will give u all the factors that will help u to fix it, and u need to work on it one after one in care:
1) Disable x-windows, gnome, ... at least for all testing. This is very important to be done.
2) Disable all devices not used, USB, Video, etc. (If possible to be done), but at least you should disable the x-windows.
3)Give lower IRQ for the digium card (for higher priority).
4) echocancelwhenbridged=no in zapata.conf, bcz this can cause problems when not needed, only to be used as a last
resort for echo issues.
5) Raise gain in hardware, return to 0 in software.
In hardware, the file (/etc/sysconfig/zaptel and /etc/modprobe.conf).
6) Run fxo tune and lower the gain in software, this will remove the static sound on the line (static noise).
fxotune (type man fxotune to read about it).
fxotune -i -vv -b 3
-i is the configuration mode
-vv for vesibility
-b for testing at module 3
Note: asterisk should be stopped before running the fxotune.
6) set opermode=KUWAIT (ir country) in /etc/sysconfig/zaptel and /etc/modprobe.conf
If I am in ur case, I will disable x-windows and then I will make gain = 0 in the software and increase it only in the hardware. Also, I will set the opermode="my country". Do not forget to stop asterisk and run the fxotune to remove the statc noise.
Looking to hear from you if that problem resolved.
Regards
Bilal
> Dear Sir,
>
> I have the following Scenario:
>
> 1- I have a DID number from Voxbone mapped to my asterisk server
> with RFC 2833 protocol used for DTMF
> 2- On asterisk Server I configured an incoming peer that receives
> calls from VoxBone and send calls to a2billing context as follow:
>
> sip.conf
> [sip_proxy1]
> type=peer
> context=a2billing
> host=81.201.82.39
> dtmfmode=RFC2833
> rfc2833compensate=yes
Try adding:
relaxdtmf=yes
to the peer
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