similar to: Possibility to catch DTMF when 2 users are in a conversation

Hi, I know this question may have been asked before (although the archives don't seem to suggest it), but has anyone had any problems with Asterisk accepting a PIN number for a conference room. At this point in time I have established the conference definition in the meetme.conf file as well as specifying the appropriate lines in the extensions.conf file. meetme.conf file: conf =>

Hey! I am trying to configure the voice onference with MeetMe application for my internal users. I have my server and 4 clients on same LAN and following is my extensions.conf file: [globals] Ahsen=SIP/222 Tahami=SIP/444 Uzair=SIP/333 Wasif=SIP/555 [internal] exten => 1234,1,Macro(voicemail,${Ahsen}) exten => 4321,1,Macro(voicemail,${Uzair}) exten => 5678,1,Macro(voicemail,${Tahami})

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

What is the most efficient way to allow inbound callers to dial internal users yet restrict them from outbound PSTN calls? Today I have a basic greeting that after a welcome message allows inbound callers the ability to dial any of my users. However, it seems that since I transfer the inbound caller to a context that allows them the ability to call my internal users they have the same rights as

I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway user at a rate

Hi again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took

Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk

I am using a grandstream phone with g729 and alaw odecs and in both modes I cannot seem to pass dtmf's, neither inband nor out of band, neither wthrough a lcoal server nor through a natted connection. Am I missing something ?

Hello, Before throwing in the towel with my Sipura 3000 has anyone had much success with that adapter connected to a door phone? In our setup a doorphone is connected to the SPA's fxs port. When a visitor rings, asterisk calls a group of Polycoms and the person who answers has to enter *1 to trigger the door opening. However it seems the SPA doesn't relay the DTMF's to the

Dell Customer Communication Hi All, I am using Compiled against library: libvirt 1.2.9 Using library: libvirt 1.2.9 Using API: QEMU 1.2.9 Running hypervisor: QEMU 2.1.2 I want user to provide username and password authentication to virConnectPtr virConnectOpenAuth(const char *name, virConnectAuthPtr auth, unsigned int flags) to login remotely for the qemu

Hi, Our gateway has asked that we send DTMF as RFC 2833. Although chan_oh323 seems to do this, it doesn't specify the DTMF mode during the H323 setup headers. Is there an easy way around this? Thanks, Andrew

Hello, I am experiencing a strange problem and I am wondering if anyone may have some pointers as to how to overcome it. I have an account with VoipTalk here in the UK which I have connected to my Asterisk server. VoipTalk supports IAX2 and SIP and I have connected to my Asterisk box using both methods. The problem is when I dial into my Asterisk box via my VoipTalk incoming PSTN phone

I have the dtmfmode in sip.conf set to use rfc 2833 however, when my users have to enter pin numbers to join let say someone's conference bridge the pin is received twice. Any ideas on how to solve this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050120/56ff46fa/attachment.htm

After many hours today thinking that I had placed a bug into my dialplan, I realized that for some reason DTMF tones are simply not making it into asterisk! Calling into my pbx transmits crystal-clear audio in both directions. But dialing DTMF's from pstn->pbx is unsuccessful, while pbx->pstn works fine. The tones simply don't make it through. Tiny brief fragments are all. Please

Hello ! My asterisk log is full of messages like this: [2011-03-06 19:01:15] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:20] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:25] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer