similar to: Asterisk + Avaya DTMF problem

Hello, I'm posting this to the list in case others run into the same issue. I've recently been connecting * to a legacy Avaya InDEX switch over E1 ISDN PRI here in the UK. Everything was working OK, except that DTMF digits were not being recognised by * when sent by the Avaya switch to the * system. Instead, the background noise of the call centre would be silenced while

Hi I have a client using a Avaya IP Office PBX that is taking a SIP trunk from me terminating on a * box. It all works perfectly apart from DTMF. Although you can hear the tones they don't seem to get recognised. I have tried DTMF mode auto, inband, out of band and rfc2833 but no luck. Any ideas? Regards Jon -- Jon Farmer Telford, Shropshire, UK

Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller

Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. I played with the rx/txgain values from hearing nothing to too loud... I have no more ideas. Marc -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Steve Underwood Gesendet: Donnerstag, 4.

Hi All, I need help with integrating AVAYA 8300, the avaya can do outbound calls but cannot do inbound calls, im sending calls from sip to avaya using E1 ISDN line. My config was based on aspect dialer it's working with aspect but not with avaya. My config and error is below. zaptel.conf span=1,1,1,ccs,hdb3 bchan=1-15,17-31 dchan=16 zapata.conf group=0 context=avaya switchtype=euroisdn

Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. Any

Hello Does someones know a good low-level way to debug the DTMF that is arriving (or not arriving) at Asterisk? We've already used the DTMF logger level. The picture is the following: We are developing an application that calls to a customer to get authorization information through DTMF. We've used at development environment SIP/IAX channels and everything was great and the work was

Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is > 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue

Dear all I have recently install TE120P Digium E1 card now everything is fine and working i have connect my asterisk with avaya but when anybody transfer call from avaya i got this error on my asterisk consol [Aug 27 14:46:50] WARNING[19527]: app_dial.c:741 wait_for_answer: Unable to forward voice or dtmf -- Hungup 'Zap/32-1' I m waiting for your reply Satish Patel

Hi ..., I've a TE205P card installed in my asterisk box.Port 1 of my card is connected to service provider.From port 2 I want to connect Avay Partner system. what type of cable I require to connect the partner system (straight/cross over). How the call routing from outside will be done to epbx. Thanks inadvance regards rudra -------------- next part -------------- An HTML attachment

Hi list! I have an Asterisk server (1.2.14) connected to a E1 line via a TE410P, and some PAP2NA connected to it. The PAP2 DTMF configurations is set to INFO and Asterisk to INFO too. At first, is INFO method different from RFC2833?? Well, I have two problems. The first is that when I place a call to outside, via E1 trunk, sometimes I get some DTMF tones and I'm sure nobody hit any key. Seems

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 am using H.323 to create a trunk between Asterisk and Avaya IP Office system. Everything is working correctly, Asterisk can call Avaya and vise versa. Now I create a conference room with a user pin in Asterisk. Avaya can call into the conference room, but can enter the pin number. The error message they are getting is the "invalid pin number". However the pin number works if they

With our current setup, we have an older avaya system which is linked with our asterisk system via a em wink connection. When you press "2" on the avaya network, it will jump to our asterisk box and then sends DTMF digits. Asterisk listens for those numbers and then responses as soon as it has a match. The problem is with having a "send to voicemail" option. Right now, a user

I'm posting here in case anyone else runs into this and needs some help. I'll probably update the voip-info Wiki pages on Toshiba integration in a bit. Asterisk 1.6 makes things a bit easier than what is on that page. I'm integrating an Asterisk server with a Toshiba Strata system at my office. Right now, it is driving some VoIP phones (Cisco ATAs with analog phones plugged into

We've moved within the last two months to Asterisk 1.4.x All remote facilities are connected via highspeed (9mbit) connections (Over OpenVPN) to a central Asterisk box, acting as a voice router, that funnels all calls into our Avaya Definity G3R via PRI. When corporate employees visit the remote facilities and try to call the G3R's voice mail system(Audix), DTMF is not recognized unless

Just to add.... P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] Sending :160 bytes 2 MISDN P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0 P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0 P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] PH_CONTROL: channel:1 oad2:07nnnnnnnnn dad0:820055 P[ 1] --> DTMF

n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, --

Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which

I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems