similar to: G729 CPU Utilization

Hello; Is there CTI module in asterisk with CTI client to login and logout and do ready and pause? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130714/dd86c885/attachment.htm>

Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/attachment.htm>

Hello, I am having some problems understanding a few concepts in Samba while trying to use samba-common-3.0.0-14.3E, samba-client-3.0.0-14.3E and samba-3.0.0-14.3E on RHE 3.0. Basically, I have security = domain. My system is running winbind, I've added the winbind calls to nsswitch.conf. I can get my wbinfo -u and wbinfo -g commands to show me what I want. That all seems happy. I have a

Can't I upload the 3PCC firmware ? available from the Cisco website? Actually it came with sip88xx.... firmware. Regards . On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies147 at gmail.com> wrote: > Hi, > > You have to buy the 3PCC version for this to work. Once you have this, > they work very much like the Cisco SPA handsets. > > I also ended up with a non-3PCC

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Am looking for a service provider who can provide enterprise SIP trunk with 100 channels concurrent sessions. I see some like Inphonex, Broadvoice... and etc.... Is there any suggestions for the service providers. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to upload woth TFTP due to some reason it's getting failed. Do I need to load 3pcc firmware or anyway to Configure from the phone itself or from the GUI? I have the SEPMAC.cnf.xml as well. Any suggestions would be appreciated. Regards .

Hi All, I have a data output as below.I want to display them in an interactive html report using shiny but the data table is not rendering properly and instead giving NA values. max_usage_hours_per_region<-setNames(aggregate(df3_machine_region$sum_as_hours~df3_machine_region$Region,df3_machine_region,max),c("Region","Sum_as_Hours")) Region Sum_as_Hours 1 Africa 1156.0833

Hi All, I have a data output as below.I want to display them in an interactive html report using shiny but the data table is not rendering properly and instead giving NA values. max_usage_hours_per_region<-setNames(aggregate(df3_machine_region$sum_as_hours~df3_machine_region$Region,df3_machine_region,max),c("Region","Sum_as_Hours")) Region Sum_as_Hours 1 Africa 1156.0833

Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256

i''m running the following script: #pragma D option quiet profile:::tick-1sec / ++x >= 15 / { exit(0); } io:::start { @io_size[execname] = sum(args[0]->b_bcount); } on exit, the script prints out the value of @io_size, why? there''s no printa(), and i also specified "D option quiet" (i also tried -q). this seems to happen with any kind of probe: on exit(0) all

On 12/05/15 20:27, Greg Enlow wrote: > To the powers the might be, > > it seems that the "(root) preexec" function does not work when used with "msdfs proxy". Is that just "my" problem, an error or by design? > If it is by design, I would be curious to know why. > If it is an error, can it be addressed fairly quickly? > If it is none of the above,

Hello list, is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php Tofik Suleymanov

Hi, How to create dialplan restriction for particular extensions.. -- Thank you with regards, Gopalakrishnan A.N, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100917/a4bc96f6/attachment.htm

Hi, I was wondering if someone had any thoughts as to why "preexec" doesn't fire when "msdfs proxy" is used? Thank you, Greg Enlow -- Greg Enlow grenlow at hk.mailbox.de On 13 May 2015, at 11:18, Greg Enlow wrote: ok ok ... Names have been changed to protect the inoccent. This installation is being used to mitigate a server migration by providing read-only access

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

For the libvirt 5.8 release, I find that there is a new comment in qemu.conf: #slirp_helper = "/usr/bin/slirp-helper" It indicates that there is a slirp-helper to help setup slirp network. But I cannot find it even after I built the latest qemu(v4.1.0-1378-g98b2e3c9ab) and libvirt (v5.9.0-rc1-2-g73f91d659b). Could you please tell me where I can find that helper program? Thanks --