similar to: Inband DTMF not detected - bug or config error?

Hey all! I recently tried the dtmf_mode "auto_info" on my setup to support endpoints that only understand SIP INFO as a fallback. My setup is the following: Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO) Both are configured with "auto_info" dtmf_mode in pjsip.conf. What I ran into is, that DTMF sent from endpoint A to endpoint B is additionally sent via

Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070622/43308a1f/attachment.htm

Hello all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF: 1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF settings according to the incoming INVITE - RFC2833 or inband. The is no such settings in PJSIP. Do you know is there is a plan to develop it? 2. When we setup 2 peers, one RFC4733 and the other inband,

I've been searching to a solution to this for a while and can't figure it out, perhaps someone has done something similar. I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low on my lightly loaded switched gigabit ethernet network. One Asterisk uses Zaptel and a Digium card, and DTMF recognition

Hello Asterisk Users, I have been looking for similar auto dtmf mode implementation on pjsip, but didn't see it coming, so I decided to give it a try. My basic plan was to do it as simple as possible with minimum changes because I am not familiar with all Asterisk code. My idea is to use rfc4733 settings, but when going over the codecs check if telephone-event appear and if not set the dtmf

> I have seen the following effect in Asterisk, though: where > it converts > an inband DTMF (eg coming off a Zap channel) into an > indication, it mutes > the audio where that tone is. But sometimes it leaves a > teeny bit of the > tone behind. > > If you take such a call over say IAX to somewhere and then > back out a Zap > channel, you end up with the

Are there any known problems converting dtmf from oob over iax2 to inband over rtp/ulaw? Obviously it works when converting to inband over pri/ulaw et al, but how about rtp? I've got packet traces that confirm that 2833 packets are properly generated when I have 2833 configured for the rtp link, but the other side seems to be ignoring those packets. So I tried inband on that link; nothing

Hi, I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk..... Since Asterisk is claimed to have good dtmf recognizer, I suspect there are some settings to workarouned... I've tried dtmf relax, but didn't help, so I suspect gain settings.... Is

I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that SIP providers are not very good at doing that suppression (leaving audible clicks, or failing to suppress the tones

Is it possible to make Asterisk do inband DTMF over IAX?

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

Hi all. Why don't Asterisk support inband DTMF with G729? Is there a way to do that!? Are you using RFC2833? Doesn't it a security hole? Thanks. Denis.

Hi all, I use a GSM device to send dtmf on my asterisk system (via SIP). the codec I use is ulaw (or a-law). dtmf mode is INBAND. relaxmode is on. but most of the case, I 'missed' some DTMF or I 'double' one. as anybody as seen this before? is there any way to prevent this thanks

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi group! Does DTMF inband work with oh323 channel driver ver. 0.6.7? If yes, how to enable it, make it work? I have tried with "inBandDTMF=yes" in general context of oh323.conf, but I get this message when I * is starting. [chan_oh323.so] => (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing

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

I have many endpoints and each endpoint has some parameter in common so i wonder is there any way to config one for all endpoints? Like in my example I have two endpoints and I repeat the same thing, [100] type=endpoint aors=100 auth=100-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <100> dtmf_mode=rfc4733 use_avpf=no ice_support=no

Hi can anyone help me on this error msg?? dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames thnx St

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

I'm trying to use H323 for the first time so please forgive me if I've made a mistake here. I have downloaded and compiled the latest versions of pwlib, openh323 and asterisk. I have dtmfmode=inband in h323.conf, but the remote system is not hearing the DTMF. Running a trace reveals the following... 1:08.398 ThreadID=0x00022012 h323.cxx(4594) H323