similar to: dtmfmode=auto, but doesn't work

Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with

Hello everybody, I have been googling for hours and also searched on http://www.voip-info.org/wiki-Asterisk, but I still can not find any information for the problem I have. So I hope one of you could help me out. I have actually very little experience in Asterisk and also Linux. But by following installation guide, luckily I could get asterisk working. That is only with SIP and IAX channels

Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756]

Hello everybody, I am not using astdb (no func_db and app_db) so I am wondering why asterisk is always updating it. The interval of the update is not constant. Using lsof, I noted the intervals are somewhere between 1 minute to 12 minutes. The output of lsof says that asterisk, atd and crond processes were just active, just after the hard disk changed the state from standby to active/idle. I

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. -- Tomislav Parcina tparcina#lama.hr

All the 'unknown device' means is that your 'lspci' doesn't know what the card is. That's all. Nothing more. --Rob ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aryanto Rachmad Sent: Tuesday, 4 October 2005 7:43 AM To: asterisk-users@lists.digium.com Subject:

Hello All, Sorry to resend this email since I really want to know the answer. Did anybody know how I could make Linux Guest OS mount CD automatically with Nautilus? When I exposed a Music CD to my FC5 guest os on Xen 3.0.3, I found the CD wasn''t recognized as a Music CD, it was recognized as a blank CD. In Nautilus, when I clicked the icon of CD, a empty folder was opened. On general

Hello everyone, I have come across a bit of an odd problem: I am currently analysing data PCR reaction data of which part is behaving exponential. I would like to fit the exponential part to the following: y0 + alpha * E^t In which Y0 is the groundphase, alpha a fluorescence factor, E the efficiency of the reaction & t is time (in cycles) I can get this to work for most of my reactions,

> Date: Wednesday, November 09, 2016 21:25:50 +0100 > From: Antonio Trande <anto.trande at gmail.com> > > On 11/09/2016 09:21 PM, Phil Wyett wrote: >> On Wed, 2016-11-09 at 12:16 -0800, Bart Schaefer wrote: >>> For what it's worth, I just received the auto-notification of >>> having been kicked off the list for excessive bounces. So gmail >>>

Dear R-SIG-Debian, > > a) You still haven't explained why you need to rebuild it when > ? ? ? sudo apt-get install r-cran-rgl > ? gets you a binary I do not want to answer for the OP, but for myself, two things: 1) I maintain a personal library in my HOME folder - for a few reasons - and one of the benefits is that I can install/update R packages without needing to run R as

Hi, I''m looking for qualified Ruby on Rails developers to work on a client web portal project in Midtown Manhattan for a large financial research company. Requirement Overview: Ruby / Ruby on Rails developer with strong object oriented programming background. Good understanding of model driven architecture, MVC, RDBS and data modeling. Required Skill Set: - BS. in Computer Science (or

Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time

Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked "sip show peer" and saw that Vitelity for inbound was now reporting "DTMFmode : rfc2833" (it didn't used to), so switched my ountbound dtmfmode to

I appreciate your report (below), but it's a strange and disturbing coincidence for me. DTMF out through Vitelity was not working for me until 1-2 days ago when I changed it from rfc2833 to inband! Maybe I just missed the change date and I should change it back? ---- Date: Tue, 22 Jul 2008 12:23:39 -0400 From: "Mark G. Thomas" <Mark at Misty.com> Subject: [asterisk-users]

We have an issue with incoming calls from a provider in which DTMF tones are sometimes sent using 'inband' and sometimes using 'rfc2833'. All calls are G711 and the incoming SDP never indicates support for rfc2833. Is there a setting in sip.conf that allows asterisk to receive DTMF tones in either inband or rfc2833 formats? The option 'auto' does not work for us

It appears Asterisk can handle DTMF inband on only a limited selection of formats, of which G.729 is not one. The issue appears to be something involving "short data" -- whatever that is. (I'm inferring all this from looking at dsp.c in the vicinity of the error message I was getting, which pointed to line 1424.) What *is* "short data"? Is this really a show-stopper for

What dtmfmode should I set for IAX protocol? When I dial FWD over IAX it doesn't recognize the numbers I press. -- #Joseph

Hi, I'm am getting doubled DTMF on some digits with one of my providers who also uses asterisk. We're using SIP, with dtmfmode set to rfc2833, and the codec G.711. Once out of every five or ten calls, there are no problems, but more often than not, the DTMF is getting doubled-up on at least one of the digits of the extension dialed. I've tested with a CVS-HEAD from Febuary, and just