similar to: Asterisk QOS

Hi All, While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use it in my dialplan. I had 2 sip extensions 555 and 666 and I called from 555 to 666, but unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI. Would you please let me know what is wrong with my dialplan and/or what else should be done to get the value of ${RTPAUDIOQOS}? Following is my dialplan context

I'm having a problem pulling data from RTPAUDIOQOS. For testing purposes I have asterisk sending QOS data to the console. It seems I get QOS data only if the caller hangs up, with the variable being empty if the callee (or asterisk) hangs up. Any idea why I would see this? exten => h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS}) Thanks, Richard Seguin -------------- next part

hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval -------------- next part

I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting. This is what I was thinking of trying. 1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc). 2. The mains server will not handle RTP traffic, it will send the RTP traffic to

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

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

Hello, I have a question about best practice (or recommended practice) for allowing SIP registrations from the Internet. This is what I was thinking of implementing: 1. Use OpenSips for the SBC, enable SRTP and TLS 2. Allow limited access to the actual Asterisk PBX (behind firewall) via OpenSips Is there anything that I am missing that probably should be implemented? Thanks, Richard

Dear all, I am having an issue with CDR logging. What I want to do is log jitter variable from RTPAUDIOQOS module into Master.csv at the end of each call. I am using asterisk version 1.4.26. For CDR purposes, I am using cdr_custom, and the content of my cdr_custom.conf is the following: [mappings] Master.csv =>

hello I would like to take value RTPAUDIOQOS channel variable on DAHDI / IAX Channel... Any Idea..!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/810a0848/attachment.htm

Hi, collegues. I have installed Railsinstaller 2.0.0 and have a problem When I''m starting Command Prompt with Ruby and Rails I see the following text "The network path was not found. # Rails Environment Configuration. Your git configuration is incomplete. user.name and user.email are required for properly using git and services such as GitHub ( http://github.com/ ). Please

How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/c6d473ce/attachment.htm

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using

Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Thanks, Jim

I'm wondering if anyone knows of ISPs with service that has QoS features that would be good to use with VoIP stuff. Granted, the QoS would only be supported as long as you stayed within their network, but it might be better than nothing. --Eric

If I relay through my Cisco 7940/7960, does it do QOS, even with a dumb switch? I know you can set quality/qos but only if you have a layer2/layer3 switch that supports the tagging. A simple little linksys 5 port switch wouldn't know about QOS, it'd give everybody equal priority. If a computer plugged into the phone, and the phone into the dumb 5 port switch and then to the internet,

Hello all, Can anyone recommend a good book which thoroughly explains QoS from a Linux perspective? Something with TC examples & the like. I''ve looked at the following: http://www.amazon.com/gp/product/1580533418/qid=1148368189/sr=1-2/ref=sr_1_2/102-2819973-6353768?s=books&v=glance&n=283155 Engineering Internet QoS. Thanks.

Dear all, I want to configure QoS in my Cisco router and Cisco Switch.. Need some information. Need your help. Best regards IT Department Director of Information Technology Albert Chong 562-695-8823 Ext. 2201 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040604/ca5d22e6/attachment.htm

My router (1605R) currently does not support QoS. Is there any open source software available so that I can set one up before the router? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/38b6e3f2/attachment.htm

Hi I have Linux box (Debian) that acting as a bridge. Eth0 and Eth1 are bridged (br0). Br0 have public IP. Eth0 connects to the internet. Eth1 connect to servers in DMZ (with public IPs). Eth2 connects my Lan (192.168.1.0/24). My connections is 2Mbit/2Mbit. I''m doing SNAT for my Lan. QoS on eth0 works fine for DMZ, but is there a possibility to doing QoS on