similar to: FXO Modules and Sip Outbound

Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. _________________________________________________________________ Discover sweet stuff waiting for you at the Messenger Cafe.? Claim your treat today!

Asterisk Users, I am looking for Sip Providers for my Asterisk 1.2.13, running Debian Etch system with McLeodUSA's T1 service. Has anybody ever used Callcentric for their Sip Provider? Any service issues with Callcentric? Best Regards, John _________________________________________________________________ Messenger Caf? ? open for fun 24/7. Hot games, cool activities served daily.

Asterisk Users, I have 3 FXO modules with the TDM400P Digium Card. I can dial into the Asterisk rings my Sip phone, but dialing out with my SPA941 phone through the zap channel is a problem. I keep getting this message on the Asterisk CLI. What am I doing wrong? Thanks in advance. -- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") in new

Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John

Greetings, Are there any xorg-x11-devel or xorg-server-devel rpm for centos 5? Thanks, Barton _________________________________________________________________ Kick back and relax with hot games and cool activities at the Messenger Caf?. http://www.cafemessenger.com?ocid=TXT_TAGLM_SeptWLtagline -------------- next part -------------- An HTML attachment was scrubbed... URL:

>> I understand that that is not how scp works today.>And it will likely never change. Why not? Just because "That's how we've always not done it" doesn't sound like a very good reason to me. >> I'm suggesting that we make a minor change to how it works.>scp is maintained for compatibility reasons only, as I've understood>things. That's still

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex

Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password:

Asterisk Users, I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect on

Asterisk Users, I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service. I have two Netgear switches on my T1 router, one for VOIP and another for data. I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for all data. This morning I saw this message a few times on the Asterisk command line. The lagged cause garbled phone calls. Is my network to

Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk

How difficult would it be to add an additional parameter to the -t that would *lock* the user at that directory level. say -T instead of -t... By locking, I mean translating /path/to/file as ./path/to/file, or ../../../path/../../../path/to/file as ./path/to/file. Basically set a root point as the current home directory, then build the pathing based on that, any "../" would become

All, I have been using the following as a means of rate limiting access to the Internet via eth0 (which connects to my cable modem) and it was working great with my 2.4.20 kernel: tc qdisc del dev eth0 root tc qdisc add dev eth0 root handle 1: htb default 1 tc class add dev eth0 parent 1: classid 1:1 htb rate 486kbit ceil 486kbit tc qdisc add dev eth0 parent 1:1 handle 10: sfq perturb 10

Asterisk Users, I am running Asterisk-1.4.11 on a Debian "Etch" system. On an occasion, when customer calls into my Asterisk Box, I get this error messagefrom Asterisk "CallerID returned with error on channel Zap/3-1" , causing all my zap channels to be busy. So, I cannot make any calls in, nor out. I am located in the United States. Is there any other suggestions,

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

RF64 support sure would be nice, but it wouldn't really help to do this "instead of" CAF. For one thing, Logic Studio Pro does not seem to support RF64, because the manual states that WAVE and BWF are limited to 4 GB. CAF may be a format which lacks universal support, but RF64 is also very limited in usefulness. Treating either one as a substitute for the other is not

Avuton Olrich wrote: > On 7/11/05, Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > >>and I'm thinking of adding support for FLAC in a CAF container >>as well. Is anyone else working on this? If so please let me >>know so we can agree on how FLAC should be contained with CAF. > > > I'm sorry, but what are the advantages to the different

Hi, the CAF format is not stremable due to the way the format works. On 14 May 2020, at 4:42, webmaster at berean-biblechurch.org wrote: > Using FFmpeg, I can stream to a file on disk okay: > c:\apps\ffmpeg\bin\ffmpeg.exe -f dshow -i audio="Line In (Realtek > High Definition Audio)" -c:a libopus -ac 1 -b:a 32000 live.caf > > But, if I add Icey metadata, FFmpeg

Have some flac files, that I've been trying to re-encode again... example: abc.flac ( was coded at --fast ) and its huge so, flac -d --best abc.flac or flac --best abc.flac or flac --best abc.flac --force NONE of these work, to re-encode the file into --best file size has not change. What is the correct command line to correctly re-encode a file to --best (regardless) on what compress