search for: rtpqos

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, audio, local_jitter)}) ; the Sa...

I'm confused about something.... In Asterisk 1.4 you can collect RTP QoS metrics at the end of a call with: ${CHANNEL(rtpqos,audio,all)} Now, when your using the AMI to do a callout, like this... ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/1000 Variable: callid=849120 Variable: destination=SIP/1001 Variable: timeout=70000 Variable: timeout_warning=60000 Variable: timeout_warning_repeat=3000...

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

...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 these values in my MySQL CDR table in order to calculate a MOS value: "ssrc=592614191;themssrc=0;lp=1;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=20734;rlp=0;rtt=0.094000" codec used: g711a --...

...ter,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 seems to have had a function ast_rtp_get_quality but i cant find any information about that in sources from 1.8, only a short reference in 1.4. Channel variables like CHANNEL(rtpqos,audio,rxjitter) show only information about the local channel. So not really usefull. In some old version they seemed to have it changed from remote_jitter to rxjitter, local_jitter to txjitter and so on. Was not even documented. The 2 variables RTPAUDIOQOSBRIDGED and RTPAUDIOQOS show exactly th...

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Tariq; About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting? Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,

...en 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? I'm trying to retrieve ${CHANNEL(rtpqos,audio,all)} for Leg B. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions

...ISK-23069 - Custom CDR variable not recorded when set in macro called from app_queue (Reported by Bryan Anderson) * ASTERISK-19499 - ConfBridge MOH is not working for transferee after attended transfer (Reported by Timo Ter??s) * ASTERISK-23261 - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR variables for subsequent records (Reported by zvision) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim) * ASTERISK-23231 - Since 405693 If we...

...ISK-23069 - Custom CDR variable not recorded when set in macro called from app_queue (Reported by Bryan Anderson) * ASTERISK-19499 - ConfBridge MOH is not working for transferee after attended transfer (Reported by Timo Ter??s) * ASTERISK-23261 - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR variables for subsequent records (Reported by zvision) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim) * ASTERISK-23231 - Since 405693 If we...

...ISK-23069 - Custom CDR variable not recorded when set in macro called from app_queue (Reported by Bryan Anderson) * ASTERISK-19499 - ConfBridge MOH is not working for transferee after attended transfer (Reported by Timo Ter??s) * ASTERISK-23261 - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response (Reported by NITESH BANSAL) * ASTERISK-23255 - UUID included for Redhat, but missing for Debian distros in install_prereq scri...

...ISK-23069 - Custom CDR variable not recorded when set in macro called from app_queue (Reported by Bryan Anderson) * ASTERISK-19499 - ConfBridge MOH is not working for transferee after attended transfer (Reported by Timo Ter??s) * ASTERISK-23261 - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response (Reported by NITESH BANSAL) * ASTERISK-23255 - UUID included for Redhat, but missing for Debian distros in install_prereq scri...

...STERISK-23266 - [patch]pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset (Reported by George Joseph) * ASTERISK-19499 - ConfBridge MOH is not working for transferee after attended transfer (Reported by Timo Ter??s) * ASTERISK-23261 - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response (Reported by NITESH BANSAL) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim)...

...STERISK-23266 - [patch]pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset (Reported by George Joseph) * ASTERISK-19499 - ConfBridge MOH is not working for transferee after attended transfer (Reported by Timo Ter??s) * ASTERISK-23261 - [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response (Reported by NITESH BANSAL) * ASTERISK-23141 - Asterisk crashes on Dial(), in pbx_find_extension at pbx.c (Reported by Maxim)...