similar to: Dual dtmfmode?

Hello everybody, I have set dtmfmode=auto in my sip.conf, but that does not work and I still got the following message: WARNING[4980]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 According to http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode: auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones

I appreciate your report (below), but it's a strange and disturbing coincidence for me. DTMF out through Vitelity was not working for me until 1-2 days ago when I changed it from rfc2833 to inband! Maybe I just missed the change date and I should change it back? ---- Date: Tue, 22 Jul 2008 12:23:39 -0400 From: "Mark G. Thomas" <Mark at Misty.com> Subject: [asterisk-users]

Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked "sip show peer" and saw that Vitelity for inbound was now reporting "DTMFmode : rfc2833" (it didn't used to), so switched my ountbound dtmfmode to

Hello folks. I'm not sure if this is the right list for this question, but I'll start here. If I'm using a SIP provider and I have an entry in sip.conf that looks like: [8315551212] type => friend ... dtmfmode => inband ... When I pick up the phone, call someone through this provider, and press numeric digits to generate dtmf tones, who is actually generating the tones at the

I have done some searching and not sure this is even possible, but here it goes... **Scenario** Let's say you have an asterisk server that you use to connect to a SIP provider that you push your PSTN-bound calls to using g711 and out-of-band DTMF. The SIP phones in use are Cisco 7960's and are set to also use out-of-band DTMF. For the most part, everything works great. However, a few

Hi all, I have problem using DTMF on Snom Phones (300, 320 and 360) I read they use in preference out-of-band DTMF , and if the remote system does not support it they default back to inband. I would like to use DTMF as out of band , and I defined dtmfmode=rfc2833 in the peer configuration. Nope, I am no able to access any ouside services using DTMF; Another kind of phones, ATCOM AT320, can be

Hi, I have a setup where I have multiple lines to the same provider - in this case broadvoice. I have created the entries in sip.conf for both accounts - and independently they work fine. When they both are in use, incomming calls are placed to the one that's the last in the sip.conf file. On voip-info I found the following: *Quote:* When Asterisk receives an incoming SIP call, the SIP

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

It appears Asterisk can handle DTMF inband on only a limited selection of formats, of which G.729 is not one. The issue appears to be something involving "short data" -- whatever that is. (I'm inferring all this from looking at dsp.c in the vicinity of the error message I was getting, which pointed to line 1424.) What *is* "short data"? Is this really a show-stopper for

Hello, I am fairly new to asterisk. I am currently using asterisk as a more convenient sip side voicemail system. My problem: I have cisco 7960 phones whose out of band dtmf tones are recognized properly(when dtmfmode=rfc2833) by asterisk but whose in-band dtmf tones are recognised poorly(when dtmfmode=inband) . For example 7999 comes out as 799999, 4242 comes out as 442422 ... etc I

YUP, this is the way that asterisk works. It is going to quelch all DTMF that goes out via a SIP gateway via asterisk. I spent a long time working this through and it has to do with the way that asterisk deals with DTMF and the DSP.c module that sits inband to the RTP/audio stream. There is a flag called DSP_DIGITMODE_NOQUELCH that is broken that might allow the inband DTMF after answer to work

Hi again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took

I know I wouldn't normally want this due to double tones, but my upstream provider has an issue where they negotiate rfc2833 but then send dtmf inband. I don't expect to get both at the same time, so is there a way to make asterisk turn on both inband or rfc2833? Auto doesn't work because it sees the rfc2833 in SDP then ignores inband for the remainder of the call. Thanks.

Topology: PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server When I make a call to a VoIP user from the PSTN, the call gets routed through the PBX, and Cisco. Because of that the DTMF tones are passed inband, which I can hear on the VoIP end of the call. However, I have one extension on asterisk set up so that I can check voice mail when away from my

I'm hoping Asterisk can help me solve an unusual problem. I need two SIP endpoints (VoiceXML gateways) to transfer DTMF tones to each other. Both of them can detect DTMF according to rfc2833. However, one of them (host2) must generate DTMF inband. Happily, I came up with the following sip.conf to allow host1 to detect DTMF tones generated by host2. [in] type=peer host=host1

Thanks very much for the reply Joshua! So I guess that setting dtmfmode=auto would be the safest choice in order to strip out the DTMFs from the recording, right? Cheers! Patrick Wakano On Tue, 1 May 2018, 19:36 Joshua Colp, <jcolp at digium.com> wrote: > On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > > Hello list, > > Hope you are all doing fine! > > >

Hello all, I am trying to get voicemail to work for sip phones using g729, yes I did buy the licenses. I can get it to work using other codecs like G711 and dtmfmode=inband. But when I make a call using g729 I get "Apr 29 09:47:14 WARNING[1209214400]: dsp.c:1452 ast_dsp_process: Unable to process inband DTMF on 256 frames" on my console. I can't seem to get anything to work when

Hello, I have one problem and I need to disable dtmf (disable rfc2833, info and inband) on one (other peers must support dtmf) SIP peer . Is it possible? Workaround would be use g729 codec with dtmfmode=inband. Maybe there is better solution? Thanks for help. -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: