similar to: G.729 codec in pass-thru mode

Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include <filename>) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. Regards, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: "xxxxxxxxxx" <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx>;tag=as23a29r59To: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060>Contact: <sip:xxxxxxxxxx at

Hello, I'm using AGI scripting with asterisk and need to execute certain commands just after Dial(). But once dial command is executed, further commands/instructions are ignored. $agi->exec("Dial","SIP/100"); $dialstatus = $agi -> get_variable("DIALSTATUS"); if($dialstatus[data]=="ANSWER") { do something.......

Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111214/b462516a/attachment.htm>

Hello, Running one asterisk server with below details. Only SIP to SIP calls. No real time configuration, no recording, no voicemail, no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150 to 200 concurrent calls and we start getting problem in call quality like delay in connectivity, voice breakage etc.... Hardware: 2 Physical processor Intel(R) Xeon(R) CPU

Hi, Asterisk 1.6.2.9 PHP 5.3 Mysql 5.0 Can anyone suggest hardware specification for 500 hundred concurrent SIP only calls, no codec transcoding, no IVR, no Voicemail or so. Just plain switching. There is only one requirement is to execute one php script on call hangup (h extension) which will do some calculation and update the CDRs. Thanks, Kamlesh -------------- next part

Hello, voicemail password is not getting changed through phone handset while IVR indicates that password has been changed. During google I found that uniqueid column must not be changed so it is not changed. Please guide on this. During debug log I found below but in mysql db new password is not getting updated, [Jun 15 13:54:07] VERBOSE[6418] file.c: -- <SIP/123-00000005> Playing

Dear all, I have Trixbox 2.6 (Asterisk 1.4) installed in my voip server. I have the following short questions about the usage of G.729 codec: 1) Does Asterisk have installed the G.729 codec by default ??? 2) If I don't want to pay for a codec license, using Asterisk in "pass-through" mode for G.729 voice communications, do I just have to download the open source version of the G.729

Hello, Asterisk version 11.13.1 I'm trying use realtime meetme application with c and q option. If both options are used together in meetme table under 'opts' field, c option (Announce user(s) count on joining a conference.) doesn't work i.e. system doesn't play user counting to caller. Is it bug or some configuration problem. Thanks, Kamlesh --------------

looks like experimental/work in progress support: https://reviews.llvm.org/D62731 On Tue, May 26, 2020 at 10:39 PM kamlesh kumar via llvm-dev < llvm-dev at lists.llvm.org> wrote: > > > On Wed, May 27, 2020 at 11:06 AM kamlesh kumar <kamleshbhalui at gmail.com> > wrote: > >> Hi Devs, >> going by this link https://llvm.org/docs/LangRef.html#floatenv >>

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of

Hi Devs, I am Kamlesh Kumar,CS undergraduate at NIT Manipur,India. I have been programming in C/C++ for more than 3 year. I have gone through various LLVM Core libraries and it's tool as well implemented it in my project .I am well aware of core concepts related to LLVM and it's code base, I have read three books available at Safari Online Books 1. LLVM - Essentials 2. LLVM - Cookbook 3.

As of a few minutes ago, the Digium FTP servers at ftp.digium.com contain a new set of Linux X86-32 and X86-64 G.729 codec binaries, along with a new registration utility. The new codec binaries were produced using GCC 4.1, and are more highly optimized than the previous versions. In addition, there are now versions for both Asterisk 1.2 (and previous releases) and the soon-to-be-released

As of a few minutes ago, the Digium FTP servers at ftp.digium.com contain a new set of Linux X86-32 and X86-64 G.729 codec binaries, along with a new registration utility. The new codec binaries were produced using GCC 4.1, and are more highly optimized than the previous versions. In addition, there are now versions for both Asterisk 1.2 (and previous releases) and the soon-to-be-released

Hi, I am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

Dear sir, I am a new user of R statistical package. I want to perform adf.test(augmented dickey fuller test), which packages I need to install in order to perform it. I am getting following message on my monitor. *x<-rnorm(1000) > adf.test(x) Error: could not find function "adf.test" *I am waiting for your response. Kamlesh Kumar. -- Kamlesh Kumar Appt. No. - QQ420,

Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050209/8cdbeeec/attachment.htm

GCC's behavior matches LLVM. so should we leave it? On Tue, Dec 10, 2019 at 12:54 PM David Blaikie <dblaikie at gmail.com> wrote: > What does GCC do? > > On Mon, Dec 9, 2019 at 10:25 PM kamlesh kumar via llvm-dev < > llvm-dev at lists.llvm.org> wrote: > >> Hi Devs, >> >> consider below testcase >> $cat test.c >> __thread int mtls=1;