similar to: RTPAUDIOQOS - Depending on who hangs up the phone, it's empty

I asked this on the list over the weekend, and likely missed a few people inboxes. 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 =>

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

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

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, 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)}

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

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

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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 list, I need to set Voice QoS and SIP QoS for YeaLink. The possible values are 0 ~ 63. With Grandstream I can fill in DiffServ 46, which is EF. That's what I want. With Snom I fill in 184, which corresponds to EF or DSCP 46 (according to their wiki) But what value do I want to fill in with this YeaLink ??? This is a conversion table :

> In my case, I have good incoming quality and terrible quality going out. > That is, I can hear people perfectly well but they complain that my > voice drops out and is garbled regardless of who places the call. This suggests to me that you may have congestion problems in your "upstream" traffic flow. Setting QoS on the packets may not help, if whatever router you are using

I am thinking about a change to our company's phone "layout" and would like to get comments from people who have done something similar. Currently, we have 3 locations - each with their own Asterisk PBX. The corporate office has a PRI. Each remote location has a SIP provider for 5 channels of SIP going to their own PBX. Interoffice calls use the PSTN. Most inbound calls come to

Anyone got any tips on improving sound quality on soft phones running under Window XP SP2? I have tried Xlite, SJPhone and Firefly. They all seem to have significant sound quality problems. We have a reasonable sized network of several hundred devices connected together using Layer 2 switches, i.e. pretty dumb switches with no QoS. I also have a Grandstream connected to the same switching gear.

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

Hello, I have problems with my Samba servers. The PDC in the Windows domain is a NT4 and has the patch KB828741 installed on it. 1st server : HP-UX 10.20 with Samba 2.2.8a ------------ workgroup = FR-MON netbios name = TIMIX security = domain encrypt passwords = Yes password server = * All shares are OK when accessed from Windows workstations. But from

Cisco phones act a as a switch. If you do not use the CDP protocol to "tell" the phone it needs to be in a special VLAN (802.1q) then it will just use the access port settings on the switch, and, also allow the PC connected to the 2nd Ethernet port to have access to the network. However, if you have an all cisco powered network, with all cisco phones, I could advise you to use the CDP

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

Has anyone figured this out yet? Lots of places say to add the following to sip.conf of an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ;

Hi all, I have seen a lot of message talking about asterisk crashed when using queue and mixmonitor together. I do use both in our system and also get the crash (segfault) randomly. I don't know it is related to the reason above as I have no idea about how it happened. I get the core dump below. If anybody has any idea about the root cause of the crash, please tell me. Asterisk 1.4.13

Hi All How to have access to the IAX2 call statistics inside the dialplan (not CLI)? I have no IAX2 clients (yet) to test, but do RTPAUDIOQOS.* variables do the job? Are they available to IAX2 calls as they are for SIP? Stats like total packets sent and received, lost pkts, rtt, etc. would be nice. cheers Ethy