search for: dtfm

My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it works in version 1.0.5 (was working with 1.0.3). I'm using SPA-3000 and dtmfmode=inband -- #Joseph

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...ays dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed Asterisk so that it is possible, using a PIN code, to dial out from Asterisk onto the local PSTN. This also works occasionally. Looking at the message from the Asterisk box it is clear that sometimes numbers are missed or repeated in the dial string. This I...

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...

Hello, I am sorry that I post questios regarding Broadvoice here, but unfortunalelly their support is very very bad. The simply do not answer to any emails or telephones. Last week something happened to their system. I was not able to receive incommming calls etc. Now it is back, but the voice quality is terribe and the DTMF is not working.(Is the inbound mode the correct one?) Does anybody knows

Hello,Maybe I don't understand what DTMF in ASCII means but I can't make my record stop using this syntax in a PHP agi script :fwrite(STDOUT, "RECORD FILE /var/lib/asterisk/ENR/jeanpaul wav '#' 15000 BEEP s=3000\n");The php syntax isn't a problem because I really start recording, I have a beep, the record can't long more than 15sec and after 3sec of silence my

Hi Gang Not a specific Asterisk Question. But I wonder, if the called party replies with 183 + SDP indicating support for telephony-event. Should the caller be able to send DTFM Tones? Swiss Railways uses an IVR that kicks in before the call is answered. So far I have found no SIP Phone which would allow sending RFC4733 during the early audio phase (so I cannot test if Asterisk would forward them) rendering the IVR unuseable. But the RFC itself suggests that there is no...

...ey don't seem to work with MeetMe. I've made a simple one-digit entry that calls "Verbose" to output something to the CLI. It works fine in a normal call but it doesn't work when the call is in the MeetMe room. I looked for Asterisk manager events that might get fired when DTFMs get pressed. Unfortunately I found none. Do such events get fired? If so - how do I enable that? What I'm asking: Is there a way to receive DTFM digits asynchronously? Or to get "features.conf" appmap's to work in a MeetMe room? Or to get Asterisk to fire manager events when...

...xtension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing "1" from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my "1" as "11" ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) & via SIP INFO [Dec 31 10:45:40] WARNING[17928]: pbx.c:2518 __ast_pbx_run: Invalid extension '33', but no rule 'i' in context ...[snip]... Same problem when pressing "3"... Thank you. Jonas. -------------- next part -------------- An HTML attachm...

Hi! I'd like to ask whether someone did test Opus in real-world VOIP (SIP). Did someone e.g. some characterization about sending faxes or DTFM through Opus? Does it work and if yes for which bitrates? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20130518/907c5cbf/attachment.htm

.... The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The phone quality of the spoken call is fine, but DTFM tones aren't working. I'm using ulaw as the codec and bandwidth has been set to high in iax.conf. Any advice would be great. I could post debug logs of a call if someone would care to explain exactly what to capture. I'm still a newbie to Asterisk. Thanks in advance. -- Robert P...

...#39;s sip.conf "rfc2833" DTMF mode, at least as of CVS-v1-0-03/23/05-21:40:48. When I get more time, or if someone else has the time, an examination of what changed to cause this could enable us to fix the heart of the matter. Other users on the Asterisk list (see thread "*-1.0.7 DTFM => Not working" from 03/23/2005) have reported other UAs not working. Therefore, there may be a bigger problem with the fundamental issue at hand: when do we change DTMF in channels, to ensure compliance with standards, as well as compatibility with older UAs. Hope this helps someone. S...

...enu here */ } ---------------------------------------------------------------------------------------------------- I guess to use the pbx_builtin_background ( that already implement a loop playing waitting a digit ) to play the menu , and allow Admin/User choose a option sending DTFM. And i would like to know the better way to implement that ... any hint about ? Thanks a lot, P.S : Future plans will be more complex as be able to join a new caller to conference room using outgoing call , implement the options of Admin/User menu as : - Give/Remove Talk only...

Hello, It's possible to provide threewaycalling service in asterisk (nor in terminals) for SIP users? I would like to be able to join to calls in a threewaycall sending some dtfm. Thank you in advance for the information G. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040527/ab63ea34/attachment.htm

...Zaptel 1.4.0. I want to include a special route when a certain caller calls into via PSTN. The problem is that I cannot detect the Caller ID. I tryed various setting (cidsignalling, cidstart) in my zapata.conf, here is the last version of it: group=1 signalling=fxs_ks usecallerid=yes cidsignalling=dtfm cidstart=ring hidecallerid=no callerid=asreceived language=en context=zap-incoming channel => 1-2 If you know how to aquire UAE Caller ID with this hardware, please help me. Cheers, Mischi

...-- Stopped music on hold on SIP/zytek-087a2000 -- Started music on hold, class 'default', on channel 'SIP/zytek-087a2000' -- Stopped music on hold on SIP/zytek-087a2000 == Spawn extension (firma, 113, 1) exited non-zero on 'SIP/zytek-087a2000' Asterisk 1.4.7.1 Maybe dtfm ? My gateway in on rfc, office is on info, but problems are same here and there. -- .: Jakub G?azik, .: email & jabber: zytek<at>nuxi.pl

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...

Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Can anyone recommend the Server Specs that is ideal for this scenario. Im planning to lease a server. Calls are purely SIP or IAX2 only. Thanks in advance.

...=> SIPGATEID,6,Hangup exten => 000,1,DISA(no-password|default) The problem is that when the bgsound is playing, I dial 000 on the 7960, and the bgsound keeps playing. This also happens when the 7960 is in my office, hooked up to * as a local extension. I have tried all three out-of-band DTFM settings on the 7960, with no change. In my sip.conf, the sipgate account is set up with "dtmfmode=info" which I thought might have been causing a problem until I tried ringing the sipgate DID from my mobile, which just worked. Typical. I'm starting to think that there must be ano...

Hi We are about to buy several Snom phones. Does anyone have warnings or advices against these phones ? Our finalists were Cisco, Polycom and Snom. We will be using only the SIP protocol. Thanks Patrick