similar to: Does anyone know how to get rid of this warning message?

Hi, I recall seeing a sample extensions.conf file that allowed tollfree calls to be routed via iaxtel to the US and the NL, but I must be going blind, because I've scoured the list but can't find it. Can someone send it to me if they have it? Much appreciated. Thanks! --- Paul Cheng M?ty?s kir?ly ut 10 H-1121 Budapest HUNGARY paul.cheng@alum.mit.edu mobile: +36 30 381-9311

Hi, Does anyone know the difference between RFC2543 and RFC3261? They are both SIP, but apparently incompatible. We are testing some hardware devices that support RFC3261 and it appears Asterisk is supporting RFC2543 and not completely compatible with RFC3261 (see below). Does anyone know how to configure Asterisk so it can do what is missing? 3. The problem occurs with confirm message

hi there tried to use subset with drop=TRUE, but all the 'old' levels are preserved, i.e. when calling e.g. ftable a lot of zeros are displayed >x<-subset(LREG, (kir=='AA' | kir=='BB') & (type=='t1' | otype=='t2'), drop=TRUE, select=c(event, kir, type)) > ftable(x) i explicit have to call factor like

Hi thanks to everybody who responded to my earlier post. I have looked at all the material and links provided and tried everything in there, but it simply won't work for me. My SIP phones register with Asterisk, but they cannot be called (everybody is busy at this time) nor can they call anything (error code 4, whatever that means) not even internal (yes I did give them appropriate

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 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 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

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 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

Happy New Year to everyone! I'm trying to track whether this bug: http://llvm.org/viewvc/llvm-project?revision=163923&view=revision has been fixed in the Apple LLVM version we use: Apple LLVM version 4.2 (clang-425.0.28) (based on LLVM 3.2svn) Any ideas how I can check that? Thanks, Ákos Somorjai Developer Support Manager GRAPHISOFT | Graphisoft Park 1. Budapest 1031 Hungary | +36 1

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

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, I am not an professional system admin, but I am running a small debian mail server for many years. It is time to migrate from mbox to Maildir. There is a nice and clean description for doing this without service downtime at https://wiki.dovecot.org/Tools/Doveadm/Sync I have to set per-user config for mailbox type using userdb extra fields. However, I miss some information and I can not

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

OK I did some researches and tests with it, and finally: I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console: WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames My conclusion is that the snom100 utilizes MSGSM