similar to: dtmf detection not working on sip trunks using asterisk-1.4.15

> I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in > routing calls to upstream carrier via SIP trunks out.? I spent a lot of time > in the lab testing 1.8 which included heavily testing DTMF with no issues > that came up.? It all just seemed to work fine.? But then again you can?t > reproduce every real work scenario in the lab. > > > > I?m

Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP trunks, and a second (PBX) which has UAs registered to it. We have a customer that has an existing pbx that we trunk analog lines to using a GXW-4008. The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF. The issue I'm

Hi All, I'm using 1.8 Asterisk and i havet set DTMF mode=rfc2833 in SIP global default settings. but when user sending DTMf event with SIP info method my asterisk accepting that DTMF. If default or global setting is rfc2833 then how come asterisk accepting SIP info dtmf event? what to check please guide Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone see this before? I have a main Asterisk box 11.4 connected to Windstream via SIP trunks in my colo. So as a did comes in they are routed to appropriate customers, in this case another asterisk 11.4 box. All is working well with the exception of DTMF. Losing the last digits so say someone hits 123... on the customer box I only get 12 This is the weird part, it only happens on 1 DID. If I

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

I have a telco providing DTMF inband, they say they can't provide it any other way. This is creating headaches for me. What is the common method for SIP DTMF? Kpml, or 2833 or inband? My handsets don't support inband so I'm tying up some expensive resources to convert the inband DTMF to out-of-band DTMF... Can you recommend a vendor in US that provides SIP with DTMF in RFC

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

Hi, I have encountered a DTMF issue. My scenario: Access carrier-----sip----> Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind

Hi, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using

Hi. I am having problems with asterisk detection dtmf properly in asterisk 11. I am up to rev 390229. Now, when coming in off a did we have with Velocity, the dids work fine, but from extensions often it misses digits -- I can type *4 and it will miss the 4. Often, if I type quite slowly things will work properly. All dtmf modes are set to rfc2833. Strangely enough, I did not notice this with

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Hello I have a setup where i have 2 asterisk servers connected over the public internet with plenty of bandwidth, NAT on one side only. If i use IAX between the two *'s dtmf is flawless. If I use SIP, DTMF detection is around 30% or less. I have an exten to dial into and check DTMF: exten => NPANXXxxxx,1,Answer(); (actual number blanked for privacy) exten =>

Hi all I have posted a question on the asterisk dev board about this issue but I want to see if any users have run up against this. This issue is that when calls are run through Broadvox and Level 3 the in-call rfc2833 dtmf is not reliable. This occured for me on asterisk version 1.6.1.18, 1.6.1.20 it appears to have been fixed when I went to 1.6.2.11 but broken again in 1.6.2.12-rc1. I have

[moving to asterisk-users by request] On Tue, Jan 27, 2009 at 12:56 AM, John Todd <jtodd at digium.com> wrote: > > On Jan 26, 2009, at 7:38 PM, James Lamanna wrote: > >>> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote: >>> >>>> Hi, >>>> Is it just me, or does DTMF queuing not work properly? >>>> I'm consistently faced with

We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones). The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that

I realize I cannot use inband audio for phones (voicemail and internal ivr, password for external trunks and other thing not working) So I put everywhere rfc2833. Doing this, anyway, make any EXTERNAL IVR NOT working. I see a lot of posts about this, but no solution, becouse using inband audio (which works for outside...) breaks inside IVR Is it possible to define to use inband audio ONLY on

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, When making a call like the one below, I get double DTMF tones on the PSTN side. DTMF tones sent from the PSTN arrives squelched on the SIP side. SIP > Asterisk2 > IAX2 > Asterisk1 > ZAP > PSTN SIP has been configured to use rfc2833 on both the SIP endpoint and the Asterisk. SIP endpoint also suggests a payload value of 101.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 It seems like Asterisk are having problems detecting DTMF digits when using an Adit 600 channel bank via MGCP. I've tried to turn on RFC 2833 on both Adit and Asterisk, but no digits at all are working then. Anyone experienced simular with Adit or other channel banks? I'm also unable to use V.90 modem through my setup (Adit600 via MGCP -

Hello All, Does anyone knows a good carrier who can pass DTMF tone while doing Call Back? Currently, the Call Back system works within US, but as soon as international users tries to enter phone number the system does not understand the tones. I tried to change the sip config to inband, auto, RFC2833 but it didnt work... So I suspect its my VoIP Carrier who doesn't pass the

Hi all, When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, at the end of the recording. I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards: a TE420 w/Octasic and pri_net