[asterisk-users] Asterisk 1.8 - RTPAUDIOQOSLOSS does not show audio quality

Hi all, I have upgraded Asterisk from 1.6 to 1.8 I would like to measure the quality of the audio and I want to check lostIn packets and lostOut packets as I did in 1.6. Now, I get the following for every test I perform

Re: [asterisk-users] Play audio file for both Caller and Callee in a call

Hi all, I made it easier, AMI was not required, it can be solved directly in the dialplan: same = n,Dial(SIP/${TRUNK}/${ARG2}${NUM},60,M(inject^${CALLERNUMBER})) [macro-inject] same = s,1,Originate(Local/trunk@injectWarning,app,Playback,Message-Callee) same =

[asterisk-users] ChanSpy in whisper mode - low quality audio

Hi all, I succeed in injecting audio into one channel by mean of ChanSpy, but this audio cannot be listened correctly. I am using softphones for Android and iPhone and there are so many cuts so that they cannot understand what is said in the audio file. Is this because the total RTP bandwidth

[asterisk-users] Play audio file for both Caller and Callee in a call

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! Este mensaje

Re: [asterisk-users] how to know RTP por of a SIP client in

AM, ISABEL ORDAS ARNAL i...@tid.es wrote: Hi all, How can I get the RTP port one SIP client is using for sending/receiving RTP flow? Can I obtain it in from SIP_HEADER of something like that in the dialplan? Thank you! ** ** Isabel Este mensaje se dirige

[asterisk-users] how to know RTP por of a SIP client in the dialplan

Hi all, How can I get the RTP port one SIP client is using for sending/receiving RTP flow? Can I obtain it in from SIP_HEADER of something like that in the dialplan? Thank you! Isabel Este mensaje se dirige exclusivamente a su destinatario. Puede consultar

[asterisk-users] Asterisk dialplan macro output

Hi all, Is there a way to read in the dialplan a macro output parameter? For instance, in the following macro I would like to know the pid of the Linux process for killing it when hanging up. [macro-capture] exten = s,1,NoOp(Caller IP = ${ARG1}) exten = s,n,NoOp(Filename = ${ARG2}) exten =

[asterisk-users] Monitor does not work well (little cuts in the audio file)

Hi, I have been testing MixMonitor and Monitor to record some calls in Asterisk and I have noticed that MixMonitor works fine whereas in the Monitor files of the 2 separate channels, we can find little cuts of the audio. We are using U law codec and wav files for the recording. Anyone have

Re: [asterisk-users] Monitor does not work well (little cuts in the audio file)

of my testing, using MixMonitor and monitor at the same time. Everything worked perfectly well no issues even on Vmware. Can you check if the CPU utilization is normal. Also which version of asterisk you are using? -- Regards, Sammy On Thu, Oct 20, 2011 at 1:12 PM, ISABEL ORDAS ARNAL i...@tid.es

[asterisk-users] RTP ports used by Asterisk in dialplan

Dear all, Do you know if there is a way to know the 2 RTP ports that Asterisk is using for audio flow in a call in the dialplan? I would like to launch a Linux shell command tcpdump to capture audio flow in those 2 RTP ports before call starts and stop capturing at the end of the call.