search for: 2833

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 19:01:30] DEBUG[20556] res_rtp_aster...

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...

...#8 in the softphone, I can hear the two digit and inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli output in this case is: [Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' received on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read:...

...#8 in the softphone, I can hear the two digit and inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli output in this case is: [Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' received on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read:...

...#8 in the softphone, I can hear the two digit and inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli output in this case is: [Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' received on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read:...

...#8 in the softphone, I can hear the two digit and inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli output in this case is: [Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' received on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read:...

On Wed, Feb 4, 2009 at 4:00 PM, Gregory Boehnlein <damin at nacs.net> wrote: > Hello, > Is anyone running Asterisk 1.4 w/ RFC2833 to Level3's SONUS network? > We are unable to get reliable RFC 2833 DTMF working, and have instead had to > use G711ULAW w/ INBAND DTMF to get around the issue. Looks like an issue on > the SONUS side. > > Anyone else have this issue? > Welcome to the club! ;) I'll be bl...

Hi folks, while chasing down a logging-related situation, I happened to notice that when I connect via ssh to my system it makes the following logs: Nov 15 14:15:39 saturn sshd[29868]: Accepted password for dave from ::ffff:10.0.10.14 port 2833 ssh2 Nov 15 09:15:39 saturn sshd[29867]: Accepted password for dave from ::ffff:10.0.10.14 port 2833 ssh2 Nov 15 09:15:39 saturn sshd(pam_unix)[29869]: session opened for user dave by (uid=0) That is, sshd appears to log the connection with GMT (or UCT) before the child sshd logs it with local tim...

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

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 2833? ----------------------------------------- Disclaimer: This e-mail...

Hello I have two data frames, SNP4 and SNP1: > head(SNP4) Animal Marker Y 3213 194073197 P1001 0.021088 1295 194073197 P1002 0.021088 915 194073197 P1004 0.021088 2833 194073197 P1005 0.021088 1487 194073197 P1006 0.021088 1885 194073197 P1007 0.021088 > head(SNP1) Animal Marker x 3213 194073197 P1001 2 1295 194073197 P1002 1 915 194073197 P1004 2 2833 194073197 P1005 0 1487 194073197 P1006 2 1885 194073197 P1007 0 I want these two d...

Hi, fp is a data frame like this ,----[ fp ] | Frequenz AmpNorm | 1 3322 0.0379490639 | 2 3061 0.0476033058 | 3 2833 0.0592954124 | 4 2242 0.1275510204 `---- i want to find the "Frequenz" where "AmpNorm" is max. I use this line as workaround: PeakFreqHz = subset(fp, AmpNorm == max(AmpNorm))$Frequenz[1] Is there something nicer? And is there an easy way to do the same on "predict()...

...managed to get X-Lite (v2 build 1050) working pretty well with *, but am having problems with the DTMF signalling. I've used inband signalling with no problems on the uncompressed codecs (G711), but obviously this doesn't work with the compressed ones (GSM). However when I try to use RFC 2833 it doesn't seem to pick up "0" properly. For example if I dial the voicemail app and then enter my extension (600) it says "Login incorrect" even though I haven't been prompted to enter my password, and the console shows the dialed number as 6 or 60. Has anyone else got...

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

...IP phone (find here: http://www.tootoo.com/d-rp20207560-VoIP_phone/) on an Asterisk 1.2.26-BRIstuffed-0.3.0-PRE-1y-q version of asterisk. I am running a C-based prepaid application based on MySQL that accepts dtmf events from the phone to authenticate. When asterisk is configured with DTMF mode rfc 2833 or auto (or any of the others in fact) the ast_app_getdata method sometimes just times out and doesn't get the DTMF events from the phone (the phone also has support for rfc 2833 or inband). However when using auto or 2833, my software Twinkle client will ALWAYS get its DTMF events detected wit...

https://bugzilla.mindrot.org/show_bug.cgi?id=3628 Bug ID: 3628 Summary: tracking bug for openssh-9.6 Product: Portable OpenSSH Version: -current Hardware: Other OS: Linux Status: NEW Keywords: meta Severity: enhancement Priority: P5 Component: Miscellaneous Assignee:

...n if true, G.729 (rather G.729I - 11.8/8/6.4 kbps) does not allow me to go below 6.4 kbps if needed. Basically I am looking for a single codec which would allow me to play with multiple (low) bit-rates within voice and switch to a higher bit-rate upon detecting a signaling tone instead of using RFC 2833/out-of-band signaling methods. I hope speex is the solution am looking for! Regards, Mainak PS: In any case I will have to test it out myself but a positive answer to this query would definelty brighten up the road ahead :-) -------------- next part -------------- An HTML attachment was scrubbed.....

...span t-1 card from Digium? The server is a fedora box with a dual core xeon at 2.0 Ghz and 2 gigs of Ram. Is there a rule of thumb to go by as far as conferencing resources? Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 1-800-666-2833 x299 (608) 783-7560 x299 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060228/a0495eea/attachment.htm

...10 seconds, is this the problem! I implemented the priorities within queues.conf, does that cause any problems. I would appreciate any feedback!!! Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. 1319 Saint Andrews Street La Crosse,WI 54603 (608) 783-7560 x1299 1-800-666-2833 x1299 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/1eed12e3/attachment-0001.htm

...daily)) == "Wed" ) week.dat &lt;- daily[wed.index] > week.dat A000010 A000020 A000030 A000040 A000050 A000060 2000-01-05 3925 NA 3950 6350 21717 836 2000-01-12 3500 NA 3600 6000 NA 1021 2000-01-19 2970 2701 3135 5600 21812 787 2000-01-26 2970 2616 2810 5300 20579 783 2000-02-02 NA 2833 3030 6700 21812 NA 2. If Wed observation is NA for any stock, then I want to replace it with Thurday observation. 3. If Thurday obs is still NA, then replace it with Tuesday obs. 4. If still NA, then let it be recorded as NA. Does anyone have some fancy idea to do steps 2-4? Thanks in adv...