search for: remote_jitter

...quested 'rtpqos, audio, local_jitter' - Executing [h @ macro-special1: 13] Set ("SIP/1010-00000002", "CDR (ljitt) =") in new stack [September 5 22:39:33] WARNING [14432]: SIP / dialplan_functions.c: 221 sip_acf_channel_read: Unrecognized argument 'rtpqos, audio, remote_jitter' to CHANNEL [September 5 22:39:33] WARNING [14432]: func_channel.c: 393 func_channel_read: Unknown or unavailable item Requested 'rtpqos, audio, remote_jitter' Any idea how I can fix? Best regards, Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL:...

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

...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 the things i want, but all information is stored in one field so its not really usable because it looks ugly in CDR report and doesnt show packet loss in %....

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).