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
*extensions.conf*
[a2billing]
exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten => _X.,2,DeadAGI,a2billing.php
exten => _X.,3,Wait,2
exten => _X.,4,Hangup
exten => _X.,21,Playback(AR_GetGiveToID)
exten => _X.,22,Wait(2)
exten => _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten => _X.,24,Wait(2)
exten => _X.,25,Playback(/tmp/asterisk-recording)
exten => _X.,26,Wait(2)
exten => _X.,27,Hangup
My problem is that when entring the PIN number I did not notice that any
DTMF digits has been sent from VoxBone to my asterisk server, and the IVR
continue asking to enter the PIN number all the time as you can see in the
below log messages:
----->
-- <SIP/voxbone.com-0a02e0d8> Playing
'prepaid-enter-pin-number'
(language 'en')
a2billing.php: file:Class.A2Billing.php - line:1790 - RES DTMF :
a2billing.php: file:Class.A2Billing.php - line:1794 - CARDNUMBER ::>
a2billing.php: file:Class.A2Billing.php - line:1798 -
PREPAID-NO-CARD-ENTERED
a2billing.php: file:Class.A2Billing.php - line:1780 -
PREPAID-NO-CARD-ENTERED
a2billing.php: file:Class.A2Billing.php - line:1788 - Requesting DTMF,
CARDNUMBER_LENGTH_MAX 15
-- <SIP/voxbone.com-0a02e0d8> Playing
'prepaid-enter-pin-number'
(language 'en')
a2billing.php: file:Class.A2Billing.php - line:1790 - RES DTMF :
a2billing.php: file:Class.A2Billing.php - line:1794 - CARDNUMBER ::>
a2billing.php: file:Class.A2Billing.php - line:1798 -
PREPAID-NO-CARD-ENTERED
a2billing.php: file:Class.A2Billing.php - line:1780 -
PREPAID-NO-CARD-ENTERED
a2billing.php: file:Class.A2Billing.php - line:1788 - Requesting DTMF,
CARDNUMBER_LENGTH_MAX 15
-- <SIP/voxbone.com-0a02e0d8> Playing
'prepaid-enter-pin-number'
(language 'en')
What do you think the issue could be?
Regards
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20081002/984a3537/attachment.htm
On Oct 2, 2008, at 5:27 AM, michel freiha wrote:> 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=yesTry adding: relaxdtmf=yes to the peer Fred Posner fred at teamforrest.com Tel: +1 (212) 937-7844 x501 www.teamforrest.com Using VoIP? SIP: fred at sip.teamforrest.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081002/d7901cc4/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2162 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20081002/d7901cc4/attachment.bin
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=yesTry adding: relaxdtmf=yes to the peer