[asterisk-users] DTMF not being heard correctly by far end conference system

[Thorsten Göllner]

Am 12.01.2011 11:37, schrieb Duncan Turnbull:
> Hi there
>
> I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can't nail down the reason. I don't believe this is a general issue, rather some specific conference systems that they need.
>
> I am sure I saw this covered a few years ago but can't find it in the lists.
>
> The phones and the system are using rfc2833 and either alaw or ulaw, I have stayed away from in band dtmf, but may need to consider it. They also use *1 to turn on call recording and I am not sure how that will go with inband.
>
> Another 1.6 system has no problem with being detected and it uses SIP trunks from the same supplier as the customer.
>
> The first system is a 1.4.38 box, it has sip trunks as the primary outbound route, the secondary route is iax to another box then via analogue lines. Almost all the handsets are sip and a re a mix of polycom and yealink.
>
> The sip trunks routed out through the iax link via analogue lines seem to work okay too. I am wondering if the iax handling of dtmf matches whatever the far end is expecting a little better
>
> For now I have routed everything via the iax / analogue lines which may cause some problems in terms of line availability but gets past the issue. I am considering upgrading the box to 1.6 as the working one is 1.6
>
> The other box is a digium AA50 appliance so I can't do much with it, other than find the right settings.
>
> I have on the first one
> relaxdtmf=yes			- relates to old issues too as far as I can tell
> rfc2833compensate=yes	- this only appears to matter for inbound
>
> I'm not sure these do anything useful
>
> > From what I can tell it could be the toneduration, but don't know what it should be, and while technically its probably the IVR being fussy that doesn't help me and I want to see why one system works and one doesn't
>
> This is dtmf debug from an iax handset sending digit 4
>
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-00000639 to write format slin
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 160 sample intervals
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:5297 ast_channel_start_silence_generator: Started silence generator on 'SIP/xtreme-00000639'
> [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2796 ast_rtp_raw_write: Difference is 1736, ms is 237
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 0 sample intervals
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:5310 ast_channel_stop_silence_generator: Stopped silence generator on 'SIP/xtreme-00000639'
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-00000639 to write format alaw
> [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 1713844722 to 565606422 due to a source change
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF begin on channel (IAX2/419-13088)
> [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update
> [Jan 12 23:13:55] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-00000639
> [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 565606422 to 226872656 due to a source change
> [Jan 12 23:13:56] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF end on channel (IAX2/419-13088)
> [Jan 12 23:13:56] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update
> [Jan 12 23:13:56] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-00000639
> [Jan 12 23:13:56] DEBUG[8717]: res_features.c:1399 feature_interpret: Feature interpret: chan=IAX2/419-13088, peer=SIP/xtreme-00000639, code=4, sense=1
>
> I will get a sip dump but am remote for now and don't have sip access
>
> All pointers and knowledge appreciated
>
> Cheers Duncan
As far as I can remember you should take a look at the used codec and 
this here:
http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode

Some codecs do not feel happy with some seetings for dtmfmode. Perhaps 
you may comapre these on your 2 boxes.