similar to: problem with gsm codec

Anybody has figured out why asterisk + snom have such bad quality using GSM? When I use GSM I see such messages dumped on asterisk console: WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect

Hello list! I'm using "Asterisk CVS-11/22/03-04:28:51" and try to route my normal (classic) phone calls via nikotel (www.nikotel.com). I can talk about 1 minute and get then disconnected. Here my current configuration parts which affect nikotel: register => chabrol:PASSWORD_REMOVED@nikotel/500 [nikotel] type=friend secret=PASSWORD_REMOVED username=chabrol fromuser=chabrol

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

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 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

Hi, I have a problem with asterisk-0.5.0 which I don't understand. The monitor says when making a call: *CLI> -- Executing Dial("SIP/roger-c456", "Modem/ttyI0:BYEXTENSION|60|tTm") in new stack -- Called ttyI0:1234567890 WARNING[196621]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[196621]: File dsp.c, Line 1198

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

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

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,

hi .. I have finally managed to get Chan_H323 & G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib & oH323 with versions taken from nufone's site

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

Hi, my setup is quite simple: an asterix CVS of 2003-11-15 on a 2.4.21-debian-5 GNU/Linux box in an internal network (192.168.1.0/24, asterisk is 192.168.1.10). - with a SIP phone configured as 192.168.1.190, and with its SIP server being 192.168.1.190 - with an ISDN AVM c4 i4l card on an ISDN connection with 2 channels. I try to: - dial-in from ISDN, then transfer to the SIP

I'm currently using Asterisk behind a NAT. The only service I get to work more or less (for outbound) is Iconnecthere. Everything seems to be working, but I get this: WARNING[229389]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect proce ss 2 frames NOTICE[229389]: File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 receiv ed WARNING[229389]: File dsp.c, Line 1107

Hi all, This is my first post here - I started with Asterisk a few days ago and have "fallen in love" - fantastic product. I've only got softphones connected at the moment - I'll probably order the FXO/FXS cards in about a month (and then think about getting some hardware SIP phones). Our current phone system is quite a few years old and isn't growing with us (when a single

Hi! I've been trying to use the Voicemail (and Voicemail2) applications with an SIP Phone (X-Lite, for those who cares), when I use g.711(a/u) codec, it works perfectly with inband (it detects the whole mailbox (in my case 10007)), but not with rfc2833 (in this case, it only detects 107 as the mailbox number). With gsm codec, the inband doesn't work, I guess that's due to the

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, I am trying to install chan_capi, with asterisk (cvs) on Suse 9.0, but I get the following error: ====== linux:/usr/src/chan_capi-0.3.1 # make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO

When I saw the update for Cisco Phone RTP issue I thought I would try it. Unfortunately chan_capi wont compile on this update. Can anyone recommend a good * release for Capi, Bri ISDN and Cisco 7940's SIP 6.3. Or will CHAN_CAPI also be updated ? Running Eicon Diva Bri Cards. Error: chan_capi.c:1187: too many arguments to function 'ast_dsp_process'

Hello all, I just subscribed to this list and I hope this is the correct place to ask for help with this issue. Today I checked out asterisk via cvs and I was able to compile, to install and to run it. Everything was fine so far. Then I downloaded chan_capi 0.3.3 from www.junghans.net and tried to compile it, too. Unfortunately I get an error in line 1205 when compiling chan_capi.c : too