similar to: DTMF troubles

Hello all, I'm becoming mad in trying to solve that issue. If I make a call from any of the phone here (I have some Grandstream and a couple of Snom105 - quite one of the best phones i've ever seen, this last one), to an outside IVR system, if i try to send dtmf to choose one of the IVR options, i notice in the /log/asterisk/messages this line: WARNING[43028]: Discarding too big frame

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

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 ;

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

Hi list, I'm hoping that I'm being stupid, and someone can tell me what's going on, but for the life of me I can't figure it out. (it's been a long day, and I'm now in the last 3 weeks of organising my wedding, so I hope this makes sense ;) ) When at my desk, accessing (for example) my voicemail, the dtmf tones are passed perfectly, I can enter password, change

I am having a problem interacting with a remote IVR system when the outbound call is going via SIP. The only way that I have been able to get a response from the IVR is to set dtmfmode=info in sip.conf. Unfortunately that doesn't quite fix the problem because it will still only accept DTMF input once the voice response has finished on the IVR. If I try and press anything while the IVR is

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 List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is succesfully established. Everything is OK except for the DTMF for number 4, because if the caller from mobile dial 1004 or 1014 extensions -which

I'm having an issue sending DTMF to cisco dialing this extension I should hear the dtmf tone RTP playload 101 has been sent to the cisco phone, but no audio. in the dialplan exten => 8603,1,Answer(1) exten => 8603,n,sipdtmfmode(rfc2833) exten => 8603,n,SendDTMF(1|100) exten => 8603,n,hangup() sip.conf dtmfmode=rfc2833 SIPDefault.conf I did play with all possible settings for

Hi folks, I?m using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info won?t work with Asterisk?s voicemail system. I?m using the g.729 codec for h323 and Asterisk. I?m told dtmfmode=inband won?t work with g.729. Is it possible to use

Hello, I'm having trouble working out how to send DTMF tones to an external IVR. My system has an analog phone connected to a TDM400P card, a SIP software phone (Zultys LIPZ4) and is connected to a BRI in Australia with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched with the ISDN audio patch from Traverse (which allows the card to do voice). DTMF works fine between

I have working SIP numbers with broadvoice, and just added a DID from http://connect.voicepulse.com/ . The calls answer, but DTMF is not recognized. With "iax2 debug" active pressing DTMF does nothing. Zilch. Zero. A friend tried a different IAX2 connection, and got the same results. I see the following in the archives: On Fri, 2004-04-09 at 10:12, Robert Jackson wrote: > Hey

I am using Asterisk 1.06 Stable. When I dial my Mobile Number to check Voice Mail or my Bank Account Phone Access Number, the IVR System on the other end asks me to enter *2378 to transfer to an attendant. But When I press *2378, Asterisk tells me that it cannot transfer the calls and gives an error on CLI saying Extension '' does not exist in the dial plan. What is the trick to make

Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in

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

I am having a big problem with DTMF. I have a customer using an Asterisk 1.4.20.1 system with ZTDUMMY as the timing source. The problem is that when they dial into a conference bridge or IVR where they have to enter a code they always get an error. Either some numbers are duplicated or missing. They use Teliax for calls to the USA and Protel in Mexico. Both carriers have the same problem so

I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP. I have one user that is having problems once he connects to asterisk. He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk) which goes to my asterisk IVR. If he presses a dtmf during any message, the press is ignored unless the press was a #, 0 or *. Otherwise, he needs to wait for the