[Spce-user] Loss of DTMF function

[Daniel Grotti]

Hi,
I've never heard you have to restart the server for that.
As I said, DTMF ending up in the platform are mainly handled by
asterisk, so you should change the 'dtmfmode' parameter and apply the
changes. By default it is set to rfc2833.

Other than that, there is nothing to do.




--
Daniel Grotti
VoIP Engineer


Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com

On 11/03/2015 05:10 PM, William Fulton wrote:
> Daniel,
> 
> It seems that most of our carriers only support rfc2833 for dtmf mode.
> I will change this setting in the config.yml and see if it helps.  I do
> find it curious that is has just started happening since we have been
> stable for many months with no issues.  Is there a recommended interval
> for rebooting the server to head off problems like this?
> 
> Thank you,
> Bill
> 
> -----Original Message-----
> From: Spce-user [mailto:spce-user-bounces at lists.sipwise.com] On Behalf
> Of Daniel Grotti
> Sent: Tuesday, November 03, 2015 7:56 AM
> To: spce-user at lists.sipwise.com
> Subject: Re: [Spce-user] Loss of DTMF function
> 
> Hi,
> end-to-end DTMF just pass through NGCP, so cannot be an issue in NGCP.
> Unless the devices in not sending DTMF using INFO message.
> In that case you should enable:
> 
>     allow_info_method: 'no'
> 
> in config.yml.
> 
> NGCP handle DTMF mainly when you call Aasterisk (VM), and DTMF setting
> can be changed or tuned in config.yml:
> 
> asterisk:
>   sip:
>     bindport: 5070
>     dtmfmode: rfc2833
> 
> values are:
> 
> * inband: The device that you press the key on will generate the DTMF
> tones. - If the codec is not ulaw or alaw then the DTMF tones will be
> distorted by the audio compression and will not be recognised. If the
> phone is set for RFC2833 and asterisk is set for inband then you may not
> hear anything.
> 
> * rfc2833: http://www.ietf.org/rfc/rfc2833.txt
> 
> * info: See SIP method info and SIP info DTMF or
> http://www.ietf.org/rfc/rfc2976.txt
> 
> * auto: Asterisk will use rfc2833 for DTMF relay by default but will
> switch to inband DTMF tones if the remote side does not indicate support
> of rfc2833 in SDP. This feature was added on Sep 6, 2005 and is not
> available in Asterisk 1.0.x.
> 
> 
> 
> --
> Daniel Grotti
> VoIP Engineer
> 
> 
> Sipwise GmbH
> Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com
> 
> On 11/02/2015 07:32 PM, William Fulton wrote:
>> Hi,
>>
>>  
>>
>> We are on version 3.7 of the software.  We have been stable for many 
>> months now, but just recently have discovered an issue with DTMF.  
>> Some of our endpoints use IVR to route calls, but just recently we 
>> noticed that they were not functioning properly.  We performed 
>> troubleshooting tasks at the endpoints and traced the problem to the
> SPCE box.
>> Performing the "ngcpcfg apply" command seems to restore DTMF functions
> 
>> to the endpoints for a short time, but already today I have had to 
>> restart the services twice to fix the issue.  Is there an underlying 
>> vulnerability or bug?  Has anyone else had this issue?
>>
>>  
>>
>> Thanks,
>>
>> Bill
>>
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> https://lists.sipwise.com/listinfo/spce-user
>>
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