similar to: Confused about Asterisk 1.4 RTPQOS...

I am originating a command via the AMI with this... Action: Login Username: xxx Secret: yyy ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: Local/6505551212 at LegA Callerid: 849120 Context: default ActionID: 849120 My LegA context: ----------------------- context LegA { _X. => { Dial(SIP/${EXTEN}@Provider); } } And my default context:

Hi all Can I use ssh to have remote tar files from machine A to machine B? ssh from machine A to machine B tar all files in machine B to exact to machine A Thank you ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible "Message waiting" indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks!

Every time I go to the puppet website, I get the following when I try to click on any of the links, ie: http://www.reductivelabs.com/trac/puppet/wiki/SimplestPuppetInstallRecipe Thanks! ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

Hello Asterisk-Users, I'm trying to install addons 1.2.6 on Asterisk 1.2.16 (is that OK?), but my MySQL server is installed on a different sever, so the MAKE of the addons fails with the following (truncated) error message: "app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory". Is there any way to bypass/ignore the fact that MySQL is installed separately and enable the

--- Jan Engelhardt <jengelh@computergmbh.de> wrote: > Hi, > > > there is an interesting case when the FLAC encoder (using 1.2.0) is > given simple waves. Simple waves means: I have a list of {frequency, > duration, pause} tuples that define the monophonic tune. In other > words, exactly one frequency is played at a time. > > This is the original dataset from

I have a probe: pid111::myfunc:entry { self->arg = arg0; } pid111::myfunc:return /self->arg/ { @[self->arg] = quantize(); } works great, but i''d like arg0 to be printed in hex. I''ve tried numerous approaches to do this but i''ve hit a wall every time. I''ve tried: 1) self->arg = sprintf(arg0); no, cuz dtrace has no sprintf 2) self->arg

Hi List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

I get the error while starting hvm doms. Any workarounds... suggestions.. Thanks /Jd Some thiings that might be useful are cut-pasted below. ifconfig -a udevmonitor output (as suggested in prev posts) qemu-dm log xend.log xend-debug does not seem to have anything interesting. ifconfig -a =========== eth0 Link encap:Ethernet HWaddr 00:0F:EA:53:5C:79 inet addr:192.168.12.103

My Asterisk installation crashes frequently. Since it's a random event I am not able to reproduce it so I can't say what is making it crash. Here's a snippet of /var/log/asterisk/full just when it crashes: Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo cancellation on channel 31 Sep 18 13:42:51 VERBOSE[378] logger.c: -- Hungup 'Zap/31-1' Sep 18 13:42:51

Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

Guys, Just a quick check. From what i have read in the shorewall site, intrazone traffic is allowed completely by shorewall i.e. there is no filtering or packet size limiting ,etc,etc. I ask this becos after getting shorewall up and running well, someone has complained that they cannot print pdf files larger than 100k at one go but that they have to print one page at a time. Some details;

Hello, How can I go to the Leg B channel in Asterisk Dialplan _after_ call ends? I can use Dial G option to go to Leb B channel when call is answered, but how to go here when call ends? Is here any option/function in Dial Plan? Or should I use ast_bridged_channel(chan) to get bridged channel and try to retrieve data I need from internal structures using custom c module and Asterisk API?

An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What

I am not sure if its just me, but i am able to get only local channel variables containing RTCP QOS values. The Version is 1.8.14. I want to store values of bridged channel in CDR. Phone is Cisco 7941 SIP and with sip show channelstats i see all the relevant information (jitter,packet loss) i want to get. It even calculates packet loss in %. But i am not able to store it to CDR. Asterisk 1.4

Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send

i'ld just like to have a car unit that you can put a USB stick in, a friend has one for mp3 and it is just what i want (but w/ flac support)- now that 4GB USB sticks are around you can get a lot on there in flac! > --- Brian Willoughby <brianw@sounds.wa.com> wrote: > > > are there any car stereo flac players?

Hi, Any one know why the glibc-profile is obsoleted on centos 5 (glibc 2.5)? I've checked the glibc.spec file and find that the --disable-profile option is enabled in the configuration command. So how the original profiling functions are implemented on glibc 2.5? Is there a new way to do the same/similar thing? Thanks a lot.