similar to: G.729: Typical usage scenarios

Yesterday, I started to experiment with Cisco to Cisco SIP calls using the g729 codec. According to the documentation, both the ATA-186 and 7960 are able to make use of the g729. >From an earlier e-mail, I made a change to the configuration of the ATA, changing the values: LBRCodec:3 RxCodec: 3 TxCodec: 3 The first thing I noticed was that when I did a sip show channels, the format had

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

I have seen the list of codecs for the ATA 186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodec?Configure the codec ID. * G.723.1?Codec ID 0 * G.711a?Codec ID 1 * G.711u?codec ID 2 * G.729a?codec

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

Hi, I'm using a Cisco ATA186 and iConnect to complete PSTN calls to the US. I've noticed that when I set the Cisco ATA to use LBRCodec to 0 (g.723 instead of g.729), AudioMode 0x00150015 and RxCodec, LxCodec to 0, (use g.723) Asterisk will connect to iConnect, successfully natively bridge the call and then about two seconds later not just drop the call, but terminate unexpectedly.

Kjetil Halvorsen mentioned a book call "S-Poetry in R". I checked with amazon.com and Barnes and Noble but they don't have it listed. Can somebody please give me the complete reference and where it can be obtained. Suggestion: Might it be possible to put a list of the most important and new literature about R, S, and S+ which has a direct bearing on R in the Documentation section

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has

I've got problem with connecting asterisk to CCM. Our side has Asterisk system other side CCM , ehrn i dial a number on other side channles created , connections established but nothing happend , just silence , and after some time busy tone. Sides sending ad reciving g711 codec , but we need that sides send and recive g729 (we have licenses) , if in h323 conf i try to : disallow=all ,

-----BEGIN PGP SIGNED MESSAGE----- Greetings, I've just about got Asterisk up and running and am wondering the following. Currently, I subscribe to both Vonage and Broadvoice and as such, I've got a Sipura and Cisco ATA186. Although I'm sure this is expressly prohibited somewhere in my service agreements, can I reprogram these devices to access my own asterisk server rather than

Do functions like call forwarding, do not disturb and so on work with SIP phones? I had these features working with the S100 USB device but can't seem to get them to work with the phones that are plugged into the ATA186s. Also, how do I get an extension that's plugged into an ATA186 to present caller ID? Thanks...

I got a call from an ISP friend tonight who said he is getting calls from people who are getting signed up with Vonage. Instead of sending them ATA186s, apparently they're receiving something made by Motorola. They apparently work significantly differently than the Cisco units, and there have been some problems. Anybody know anything further? Thx. B.

Will Vonage unlock your ATA for you for the $15? Or someone else? I have an ATA from them I would like to use with asterisk as well. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Thursday, April 29, 2004 8:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Vonage and * (and what

On Sun, 14 Aug 2016 20:48:04 +0100 Alex Crow via samba <samba at lists.samba.org> wrote: > > > On 14/08/16 19:37, Rowland Penny via samba wrote: > > On Sun, 14 Aug 2016 19:18:41 +0100 > > Alex Crow via samba <samba at lists.samba.org> wrote: > > > >>> Ok, lets just run through this: > >>> You have an NT4-style PDC > >>

http://www.winehq.org/site/docs/winelib-guide/spec-file says A typical spec file will look something like this: init WinMain rsrc resource.res And regarding rsrc, "If your project does not have a resource file then you must omit this entry altogether." So my mtapi.dll.spec file looks like this: init DllMain And I only manually made one since make said it was missing. But

"[...] a virtual MAC address needs to be requested for each single IP address via the Hetzner Robot and assigned to the guest NIC [...]" http://wiki.hetzner.de/index.php/Netzkonfiguration_CentOS/en#Bridged /stephan -- Software is like sex, it's better when it's free!

We are planning our office communications system using the asterisk PBX and will be using an VOIP origination and termination service such as voicepulse to do so. I know the possibilities are endless, but I'm just looking for a typical asterisk setup for a small/medium business which will use a VOIP origination and termination service. For instance, let's say we have one receptionist,

Alice Wonder wrote: > In the server environment you almost certainly are using a virtual > machine, and to use a virtual machine you create an image. Set up the > image how you want and be done with it, you can then deploy it thousands > of times and it is set up the way you need it. Who is "you"? I'm running a home server under CentOS-7, and I'm not using a virtual