similar to: Extract a MOS value from Asterisk CDR

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 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,

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've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>: > El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > > I am trying to collect enough information about an

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

Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK,

Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP

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 I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

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

Hi all, since asterisk 11 (1.6 was okay) failed the ReceiveFax-Application when it called about "Dial" and a Local-Channel. Directly from external to FaxReceive is no problem. Cut from cli: [...] [Apr 1 11:12:31] -- Executing [s at macro-redirection:85] Dial("SIP/access-trunk-00000001", "Local/0XXXXXXX40 at x-xxx-companyXXXXX/n") in new stack [...] [Apr 1

Hello all, I have a TC400B Digium card in order to deal with transcoding and I have some trouble using it, I have a timer synchronisation problem! I would be very grateful if you have any idea to help me? It seems that the card is not correctly synchronised to the system because when I speak to one side, the sound takes 5 seconds to go to the other side, and increasing, after 30 seconds of call,

Hello everybody, I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. Best Regards, -- -- Marc LEURENT lftsy at leurent.eu -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at myip.com? The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it

Hello All, MOS and R factor are the two QoS parameters used to estimate VoIP call quality. I have found that they are calculated from other metrics like jitter, latency, packet loss,...etc. But, haven't found any formula or arithmetic rule to calculate them. Do you have an idea about their formulas or an open source that calculates them. Is it possible to interpret them from wireshark.

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw

Hello, I'm using samba 3.5.11 to connect a Windows 2003 Active Directory. With cups, samba is an part of a print server used to print to windows desktop shared printers. DNS are Active Directory Integrated. Network is both IPV4 and IPV6, IPV6 for Linux and Windows Vista and above. Some times, some users are not able to print. In logs of cups, I see to thinks "Unable to connect CIFS

Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via

Asterisk Project Security Advisory - AST-2011-001 Product Asterisk Summary Stack buffer overflow in SIP channel driver Nature of Advisory Exploitable Stack Buffer Overflow Susceptibility Remote Authenticated Sessions Severity Moderate