similar to: DTMF issue with 1.8.6.0 and SIP Trunks [WORKING]

Hello, When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway, there is problem with DTMF "out-of-band". See debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May

Hello mailing list, I have been porting one of my Asterisk boxes to 1.4 and I have encountered a nasty DTMF problem. What happens is someone might come in to my IVR and enter "12345" and what will actually come through could be along the lines of "12234445". Sometimes it works, sometimes it doesn't. I had this problem with 1.2 back in November but was able to solve it

First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both set to register with the same account right now. I shut Asterisk down on

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

I have a simple dialplan. Using the read cmd, I ask caller for his passcode. If the caller is calling from an plain old analog phone, his dtmf is not heard until the prompt stops playing. SIP phones work correctly. I've trird everything I found searching the net. I've tried all the dtmfmode. I'm using 1.4.26 Currently my vitelity sip account is setup: insecure=very canreinvite=no

Hello, I have a problem of DTMF duplication. I receive call from my provider with SIP protocol. These calls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no

Hi all, I am using an asterisk-1.4.13 connected to our carrier via SIP trunk. I use rfc2833 as dtmf detection method. After upgrading to asterisk-1.4.15 our system would not detect dtmf from a caller from PSTN anymore. When investigating the SIP traffic at call initiation I realized that in the SDP message asterisk is no longer offering the telephone-event/8000 capability. So the carrier does

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

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

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

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

Hello all, I know it was on the list before but i have some questions about the Kirk IP600v3, the requested configuration files were send private i guess Does anybody have the correct SIP settings for handsets connected to the Kirk. IP600v3 I am particulair intrested in settings regarding: -Voice Mailbox -Call waiting -DTMF settings for e.g. parking an extension with asterisk functionality

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

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

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

Having big problems and for months. Our service provider (via:talk) says they are Asterisk friendly but they are not. Here are the specifics (please read the bottom of the msg too) System: Dell SM Business server 2GB RAM, Core II Processor (should be plenty) OS: open SUSE 11 Asterisk Version: 1.4.2 Asterisk GUI Version: 2.0 The system was completely set up using the Asterisk GUI with a

Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for 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

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

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