similar to: DTMF between sip trunks and PRIs

[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

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi List, I am using Asterisk 1.6.2.18. One strange problem come into my knowledge after using this version of asterisk. Without pressing any digits or key from my mobile, I am getting DTMF into my asterisk server. For getting DTMF I have use one opensourse application which gets events from asterisk server and store into database. And after that I made my own script to gets these DTMF keys and

Hi, I have a 1.8.22 Asterisk (Box A) connected to a 1.4.32 Asterisk box (Box B) through SIP. The 1.4.32 box is then connected to the PSTN through PRIs. I've noticed there are occasions where I am seeing duplicated DTMF. I've verified from the SIP trace from the phone that there is only a single '3' being pressed. It appears as though the DTMF end (without a begin) that is detected

I'm trying to use a GXW-4008 for the first time to provide simple POTS. Is anyone using it? How about samples of SIP.CONF and EXTENSIONS.CONF? Do you have advice for configuring the GXW-400x for this application? How long a local loop will it support on the FXS ports? When I started to configure the unit, I was able to connect via the WAN port. Now I'm unable to connect to

Hi, I have an 8-port Grandstream GXW-4008 V1.2A ATA converter with analog phones connected to it. They work fine except for just one "feature" I would like to modify. Somehow, each time the ATA re-registers the SIP clients or each time the device has to be rebooted for maintenance, the phones ring once. This feature can be useful as it notifies the user of the re-registration.

Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone

Hi, in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option in this particular case - don't ask. Can someone explain to me what the following means *exactly* [Feb 10 21:15:40] DTMF[2538] channel.c:

I have an Asterisk running with both GXW4008 (FXS) and GXW4108 (FXO). The FXS Gateway works perfectly, no problem so far. The FXO Gateway (GXW4108) also works fine. The configuration for local settings in Brazil was quite easy, however, I still not able to make Caller ID to work. I'm setting as DTMF Caller ID type, but still not working. Let us know what kind of problem you have, maybe I

Hi, I am trying to fix a Asterisk setup with buggy (POTS) Fax machines. The setup consists of the following components: - A Digium TE121 for connectiong to E1 ISDN - Debian box with Asterisk 1.4 - Grandstream GXW-4008 SIP ATA to which the Fax machines connect I am aware of the problems with this type of ISDN <-> Asterisk <-> SIP ATA <-> Fax machine installations, e.G.

> 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

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

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 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

I have a couple of Linksys PAP2T-NA & Grandstream HT-502 extensions that are receiving random DTMF tones on their side, but that are not heard by the outside party. I have been using Asterisk 1.6.6 through 1.6.10 and have always had this issue. I am only using SIP on the Asterisk server and all extensions and trunks are set to rfc2833; outside of this issue DTMF operation works fine.

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 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 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, we have an Asterisk server basically passing on calls using the Dial application. In the pjsip endpoint settings, the dtmf_mode is set to audio. This works with most calls. However, there is a scenario where DTMF tones don't get forwarded the way I would expect them to get forwarded. A: Caller without RfC4733 support B: our Asterisk, version 17.6.0 C: Another Asterisk, with RfC4733