similar to: g729 codex experimentation

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 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

Clicking on the "For more information, click here" link on the Digium site nice brings back up the same page I was looking at before, without any additional G.729 information that I can see. I'm wondering if some kind asterisker out there could provide us neophytes with some "typical scenarios" where that codec would be useful to us. For instance, I assume that it

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

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.

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

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

Are there any other codecs that can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030723/34e950e0/attachment.htm

I noticed that there is a soft button for transfer when you initiate a call. I pressed it, and it actually put the call on hold, although I was able to call another extension. Is that soft button functional? And if so, how do you make use of it? And if not, how does one transfer a call? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I don't make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? Kim C. Callis

Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf. I call into the system from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number. Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or

As I was working on my extensions.conf file, I started to segment calling privileges. For the everyday workers, I don't free reign to LD access unless it is business related. So I was wondering if there was a way to implement some type of accounting code to be entered before accessing LD, which of course would be noted in the CDR (however it is implemented, either comma delimited or MySQL).

I was trying to do a little searching to see if there has even been a comparison between Asterisk and VOCAL or any of the other OSS packages? "Practical Voice Over IP using VOCAL" published by O'Reilly and Associates, attempts to make a strong case about how scalable VOCAL. Of course, considering that the book is written by the makers of VOCAL, it tends to have a one sided slant.

Considering that I had a failure with compiling the latest version of asterisk because of cdr_mysql, I am going to assume that I need to have a copy of the mysql headers on the system in order to compile cdr_mysql.so. Does that sound correct? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

Hello, I have been wrestling with installing the CAPI drivers for AVM Fritz in order to use chan_capi with Asterisk. I use an SMP machine, RH 9. I have found rpm's for CAPI and AVM drivers (namely: capi4k-utils-2003.06.16-08.mungo.RH9.i686.rpm and kernel-2.4.20-8- avmfcpci-03.11.02-08.mungo.RH9.i686.rpm), but I believe that they support only single processor machines. I've already

hello i am trying to make a callback solution. client will call callback number and call is terminated. now callback server will create a call for that client. actually i have a problem in this process. that server is creating call to client (UA) when previous call is not disconnected yet. UA---------->Asterisk(callbacknumber) callis answered UA<----------Asterisk(callbackserver) call is

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 have come to realize that I don't have to have a g729a license in order to make use of an ATA-186 or 7460 connecting to another 7460. I just need to allow the codec in sip.conf. Now what ramification does that have when I dial out over one of my analog line (connected to * by a channelbank and a T100P) using my 7460 or ATA-186. The only benefit I am looking for is reduced bandwidth

Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing