similar to: Call between G.711 and GSM

Hi All, When the Caller hangup at the voice menu, the wildcard X100P didn't disconnect the calls properly and it just keep looping at the voice menu and timeout and loop again, are there any methods can fix the problems? Please help! Thanks, Randal

Has anyone tried installing * on a box with two eth interfaces which is acting as a NAT box? I have only one IP at this point and I would like to get * working without all of the NAT issues. My idea is to run * on my gateway (which is also running the firewall and masquerade services). All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the NAT screen, and will connect to the *

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hi all, There is something special I must configure in order to get the voice mssage by mail? In voicemail.conf I have: serveremail=asterisk@mydomain.ro attach=yes [default] 301 => 6535,Home Mailbox,dtoma@fx.ro I have tried to let a message for 301, but this message is not forwarded by mail. I am missing something? Thanks, Dan

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

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

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

Real Server is *not* an free product! It takes more than 500 Euro for a 50 Connections a time / Licence. Stefan Jacobi <p>----- Original Message ----- From: "Rob Burris" <robeb@keepthevibe.com> To: <icecast@xiph.org> Sent: Sunday, July 07, 2002 5:42 AM Subject: Re: [icecast] Icecast is cool, but how about video? <p>> *This message was transferred with a

rsync v3.1.0 linux v4.4.104-39-default x86_64 Found in the system log: 2018-02-22T05:02:00-0700 sma-server3 python3[31371]: backintime (sma-user3x/3): WARNING: Command "rsync -rtDHh --links --no-p --no-g --no-o --info=progress2 --no-i-r --delete --delete-excluded -i --dry-run --out-format="BACKINTIME: %i %n%L" --chmod=Du+wx --exclude="/bkp/cgate-backintime"

Just to understand what you'd call "free": Is the client that you use free as well? =) Imho not, and it's quite annoying to see this trial-reminder... On Sun, 2002-07-07 at 07:30, Rob Burris wrote: > *This message was transferred with a trial version of CommuniGate(tm) Pro* > http://licensekey.realnetworks.com/rnforms/products/servers/eval/?ulf=bas > > it's free

Hi folks. The jspeex library has classes for converting speex to pcm and vice-versa. I also have other code that converts from G.711 to pcm (and vice-versa). I want to transcode G.711 to speex, using an input stream. Can I accomplish this in one step, or must I go G.711 -> PCM -> Speex? If possible in one step, is there some example code I could look at for reference? Thanks! -- Jeff

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:

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications,

Does anyone have the C reference code of the ITU G.711 Appendix II ? -- Guilherme Loch G?es "Wave after wave will flow with the tide And bury the world as it does Tide after tide will flow and recede Leaving life to go on as it was..." - Neil Peart , Natural Science

Okay this may be a very noobie question, but I've looked around the WIKI and haven't found anything about this. I was looking at some VO-IP phones, an they list one of the codecs as g.711, this isn't listed on the * page I was looking at, is this also known as another name, or is it not supported by *? Thanks Joel Duffield Near North Business Machines www.NearNorthBusiness.com

Has anyone done anything with Asterisk using the G.711 codec? Also, is there a uncompressed option so that you could assign a single port to be unconpressed audio? Thanks! Stu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030602/118e6ac7/attachment.htm

*This message was transferred with a trial version of CommuniGate(r) Pro* Greetings, aLL. There's samba-3.0.23d, running on FreeBSD-5.3 as Win2000 AD domain member. For logging user activity on share VFS module full_audit is used (with help of syslog). Logging works well, but some errors appears in log, especially when changing ACLs on share file objects from win-clients: === Nov 30

We're trying some migration tests ... from CommuniGate to Dovecot. The migration program does an APPEND into the new mailbox: src>: a0 FETCH 2900 (FLAGS INTERNALDATE RFC822.PEEK) src<: * 2900 FETCH (FLAGS (\Seen) INTERNALDATE "29-Aug-2005 18:36:54 +0000" RFC822 {1194157} src<: ) dst>: bAPP APPEND "INBOX" (\Seen) "29-Aug-2005 18:36:54 +0000" {1194157}

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