similar to: dtmf tones during conversation

Hi helpers, I seem to have a problem of intermittent DTMF tones being played during a conversation. Eg: Extn 100 takes an inbound call and all is fine. Except, at an undetermined time the person on extn 100 will here a DTMF tone for no apparent reason (it's not the caller pressing buttons). The caller doesn't hear the tone - only the called person. The call itself progresses normally.

Hi, I tried again getting DTMF detection on my ISDN devices with dahdi going again. I used the channel debug to see whether asterisk sees the frames and detects them as DTMF. Interestingly here's what works: 1. GSM phone -> chan_dahdi g1 -> asterisk -> can_sip -> SIP phone Both the GSM phone and the SIP phone can issue DTMF that will be detected as features (transfer) 2.

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

Dear List, I have been trying to extract associations from a matrix whereby individual locations are within a certain distance threshold from one another. I have been able to extract those individuals where there is 'no interaction' (i.e. where these individuals are not within a specified distance threshold from another individual) and give these individuals a unique Group ID containing

Hi, How to accept DMTF tone during ringing mode? Its possible. Regards -Hadi.Salem

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

While I was away on vacation, buried deeply in another thread (New Asterisk bounty: SIP simultaneous), Olle E. Johansson raised a question which is close to my heart - Asterisk's management model. A management model which simply manages telephone extensions and dial plans is irrelevant to most organisations. We need a model which manages users and their interaction with the PBX. I am

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We are experiencing intermittent DTMF problems here, with the following setup: ITSP -> PIX -> Asterisk (g729, RFC2833 for DTMF). I am running Ubuntu server 10.04, but Asterisk is compiled by us and not installed from the software repository. Essentially, DTMF works for some time, but at some point it simply stops and the point at which it stops appears to be random. Using RTP debug, I

Hi, I would like to use Asterisk as VoiceMail system ... the only issue I have is with DTMF recognition. Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ... inband : works, but has some glitch ... not always good ... don't know why. rfc2833 : doesn't seem to work .. info : said to be not working

Hi, My Asterisk systems runs like a dream with mISDN, SIP and even and old Digium board. But have almost in every conversation some irritating DTMF being generated. The seems to be just as often from all trunks but are worse if noise load speaker in other end. Any good advices? Where to look for forgotten DTMF detection settings? Thank you! HB

Hello, I did this post a long time ago but never solved the problem, so i'm trying again after something like 10 months, hopefully i'll find someone that found a solution ;-) When i call an external number that sends audio before call has been answered (like some PBX of public offices do here in italy), strange things happen: I'm using chan_capi, with Early B3 active, i can listen

Hello guys, I have searched high and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code:

On 2016-11-22 07:49 PM, Pete Mundy wrote: > > One direction that may be worth exploring further is his ATA's config (or perhaps swapping it for a different model). Eg adjusting echo cancellation or line impedance settings. I have to be careful here as I auto-provison these devices and changes would propogate to every user. Echo cancellation is off. Do you think it should be on?

Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP

Folks On a lark, I am trying to set up a PXE server so I can use network boots to load new computers {some of which are virtual) at home. It's more of an intellectual exercise than a work necessity. I've read several "how to" documents, and even the one on the RedHat site doesn't help. (For example, it refers to /etc/xinet.d instead of /etc/xinetd.d, and the careless

On Thu, 05 Apr 2018 13:56:11 +0200 Pino Toscano <ptoscano@redhat.com> wrote: > On Thursday, 5 April 2018 13:45:53 CEST Tomáš Golembiovský wrote: > > On Thu, 5 Apr 2018 10:34:33 +0200 > > Pino Toscano <ptoscano@redhat.com> wrote: > > > > > When writing the OVF in OVirt flavour, add a ovirt:id attribute to the > > > OperatingSystemSection tag:

When writing the OVF in OVirt flavour, add a ovirt:id attribute to the OperatingSystemSection tag: this attribute represents the numeric value of the ostype ID, which is ignored by oVirt when parsing OVFs in API mode. --- v2v/create_ovf.ml | 202 +++++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 201 insertions(+), 1 deletion(-) diff --git a/v2v/create_ovf.ml

On Thu, 5 Apr 2018 10:34:33 +0200 Pino Toscano <ptoscano@redhat.com> wrote: > When writing the OVF in OVirt flavour, add a ovirt:id attribute to the > OperatingSystemSection tag: this attribute represents the numeric value > of the ostype ID, which is ignored by oVirt when parsing OVFs in API > mode. > --- > v2v/create_ovf.ml | 202