search for: dtmf

Hi, I have connected a TDM400P to my asterisk, I have enabled DTMF CID, the data is arriving to the asterisk but asterisk isn't interpretating it: its my full log: 1. Mar 10 16:26:03] DEBUG[8715] dsp.c: dsp busy pattern set to 0,0 2. [Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Starting simple switch on 'Zap/4-1' 3. [Ma...

Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility, but this hasn't helped at all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals do not clearly state which setting this is on the handsets. I have enabled DTMF logging and spok...

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 on Box B is being turned into a duplicate '3'. Does anyone have an idea how to fix this? Do I need to lower the re...

Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for "2" : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- <SIP/test1-00000701> Playing '/var/lib/asterisk/soun...

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...

...----------------------------- Originally posted at http://forums.digium.com/viewtopic.php?t=18045 -------------------------------------------------------------------------------------------- Hi! I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing DISA seems to prevent any DTMF detection capability when using the FXO port of the TDM400. Below, config A and B and their debug logs. In Config A I use Authenticate() instead of using DISA password since it demonstrates that it's DISA that seems to prevent DTMF detection when using Zap/1. Otherwise DISA works flawlessl...

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1) redial attempt is sufficient. Replicated from different analog phones. Troubleshooting and observations: 1. Provided external power to the TDM400P with FXS daughter cards. It did not help. 2....

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-...

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...

We are having an issue where DTMF is not being sent out right away and the tone duration is inconsistent. For a test we send a '5', then a second later we send a '9', and then five seconds later we send a '5'. If you look at the logs below you can see the first '5' is played right away, then the ...

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2...

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....

Hi Tom, Thank you for your informative and helpful reply. I had considered using the relaxdtmf setting but held off this due to not using any physical connection hardware -Asterik uses both SIP in and out from an upstream provider (Gradwell.com). Is it still possible to set this when using SIP trunks only and not physical hardware? The box does have a Digium ISDN card but the ISDN is no lon...

...till I get the above message... 2nd situation : I should be possible to transfer a call by pressing # followed by the extension, but it does not work. Although I have a call-limit of '4' and thus the peer I'm transfering to should be able to receive the transfer. [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF begin '#' received on SIP/test13-0000000b [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF begin passthrough '#' on SIP/test13-0000000b [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end '#' received on SIP/test13-0000000b, duration 320 ms [Jul 8 09:46...

I have Polycom ip-300 phones that worked yesterday but dont seem to work today (at least dtmf signalling once connected to the asterisk box) The current configuration is: [general] port = 5060 bindaddr = 0.0.0.0 context = test srvlookup = yes dtmf = inband allow = all dtmfmode=inband progressinband=no disallow=all allow=ulaw pedantic=no [202] type=user secret=xxxx context=test mailbox=20...

...segfaults right after it says pound key) 7. Segmentation fault == Using SIP RTP CoS mark 5 -- Executing [*61 at line1:1] VoiceMailMain("SIP/line1-01d646b0", "6001") in new stack -- <SIP/line1-01d646b0> Playing 'vm-password.gsm' (language 'en') DTMF begin '1' received on SIP/line1-01d646b0 DTMF begin ignored '1' on SIP/line1-01d646b0 DTMF end '1' received on SIP/line1-01d646b0, duration 190 ms DTMF end passthrough '1' on SIP/line1-01d646b0 /DTMF begin '2' received on SIP/line1-01d646b0 DTMF begin ignored...

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI "confbridge show profile user <profilename&gt...

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2> [image: Edit/Delete this post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESU...

...has. Thanks, Jamie -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tom Peters Sent: 07 July 2015 20:45 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] DTMF issue In my humble opinion, adjusting this setting will (for you) do nothing, since you don't use the dahdi channels for transport. See this discussion, which I found after I posted my first response: http://www.voip-info.org/wiki/view/Asterisk+DTMF Particularly this sentence: "Note: A...

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,t...