similar to: WARNING[1264581056]

Hi can anyone help me on this error msg?? dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames thnx St

Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Thanks, -- sudhir

Hello everyone. I am having problems getting speex support. It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install

hi all I'm being spammed by these messages in the console (see below) and sound doesn't work with today's cvs. I rolled back a week, and it works fine. In addition to the sound problems, I had to enable inband dtmf squelch on the dilnk mgcp phones. if not, each pressed key was counted twice NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process

On Tuesday 03 August 2004 12:07, Chris Shaw wrote: >Are you using the double ## transfer patch or just the regular >single # that comes with CVS? Hi Chris, Where to get the 'double ##' transfer patch? We are having the same problem and I was thinking of a similar patch. Thanks, -- sudhir

Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [root@asterisk sath]# cat g723.1 - Executing SetCallerID("SIP/-08122ae0", "1001") in new stack --

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198

Juan, I dont see people here with "attitude" or any need to shut down this list as you suggested. Contrary to your experience, I find people to be quite helpful here. Steven's comment "Start here and then ask a real question" is definitely not arrogant IMHO. He is just trying to knock some etiquette on Luan. Besides, he did give him all the answer Luan was looking for.

hi ! I have asterisk with Licensed G.729 codec enabled. Whenever I make a call using this codec a warning apears as, WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames

What does it mean ?? WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm having two problems. First - I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before>? Second thing: I get a WARNING:[1209214400]: File dsp.c,

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, if i enter a "show codecs" at cli * response with: 1 (1 << 0) G.723.1 2 (1 << 1) GSM 4 (1 << 2) G.711 u-law 8 (1 << 3) G.711 A-law 16 (1 << 4) MPEG-2 layer 3 32 (1 << 5) ADPCM 64 (1 << 6) 16 bit Signed Linear PCM 128 (1 << 7) LPC10

How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

[moved to asterisk-users, as this is not a development question] At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote: >I would like to hear from any of you who has done any kind of >benchmarking on a robust hardware that can handle large call volume, >preferably with G.729 codec involved. > >We are in the process of putting together a system that should have a >quad E1 card, G.729

Hi, I have created a heatmap using heatmap.2 having 7 clusters. I would like to extract the list of genes that are in these 7 clusters. Is there any function that can be used to extract genes for each cluster? Cheers, Sudhir -- __________________________________________________________ SAVE PAPER - Please do not print this e-mail unless absolutely necessary Being happy doesn't mean

Hello, everybody: I'm developing an application using Asterisk and a TDM-400 card. I understand the concept of the difference between GSM and WAV files when using Asterisk, but I'm not happy with the sound quality with the GSM compression. It's merely *acceptable* for a telephone call, but for anything else, it leaves something to be desired. Case in point -- if you compare the

when placing calls to the system through SIP, I got these messages, Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) any explanation for this? Thanks, -------------- next part -------------- An

Hi, I searched the archives about this, but didn't find any references. When I make an outbound SIP call, the call completes and everything is fine, but in the Asterisk console, I keep getting a huge stream of warning messages: "WARNING[1200876848]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames" I thought I saw this in a post earlier, but I