similar to: Not able to get remote channel variables containing RTCP values

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

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

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,

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 List, we are in the need to reach an external Conference-System, whos numbering system is *2nnnn*. Unfortunately *2 is the featurecode for attended transfer in our local asterisk, so the call doesn't come through. Is there a way to somehow escape the featurecode, so we can reach the external Conference? Thanx in advance, Carsten.

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

Hi, While looking for the cause of disturbance in call I found this error coming in console RTCP SR transmission error, rtcp halted Google search only shows some bug reports relating to MOH and Hold. What could cause this message? Could this be a symptom causing call disturbance? Where should I start digging to find out the reason for this error? I am using Asterisk 1.4.19 with zaptel 1.4.9.2

I've seen something similar with Adtran SIP gateways. When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is

I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the

Thanks but no Adtran here. I do think these stats are indicating an issue, I just don't know how to prove it outside Asterisk. From: EWieling at nyigc.com To: tjrlist at live.com; asterisk-users at lists.digium.com Date: Mon, 19 Jan 2015 13:55:33 -0500 Subject: RE: [asterisk-users] sip show channelstats reliable? I?ve seen something similar with Adtran SIP gateways. When a re-invite

Additional info: At the moment I am running 1.8.x but the other day I was getting the same results on 11.x Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable. Peer Call ID Duration Recv: Pack Lost ( %)

I would recommend capturing traffic outside your Asterisk server with Wireshark, then running the Telephony/Rtp/Analysize Streams option to determine if you have packet loss at that point in the network. On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote: > Thanks but no Adtran here. > > I do think these stats are indicating an issue, I just don't know how to

On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog <sgriepentrog at digium.com > wrote: > I would recommend capturing traffic outside your Asterisk server with > Wireshark, then running the Telephony/Rtp/Analysize Streams option to > determine if you have packet loss at that point in the network. > > On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote:

You could use MTR command. Its a trace route improved. Marlon Araujo > On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote: > > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or,

Hello All, I have Asterisk 1.4.5 running on a SuSE 10.3 x86_64 2.6.18.2-34 I upgraded from 1.4.2 to 1.4.5 on sunday the 24 of june and since have been getting: Internal RTCP NTP clock skew detected: lsr=1402479300, now=1402675136, dlsr=196500 (2:998ms), diff=664 I see an entry in Mantis that Russell fixed code so that this will not show when it shouldn't. Would i be correct in

Hi! I use Bria and eyebeam and it seems that asterisk doesn't send RCTP keepalives when a SIP channel is on hold. This is a known issue as is described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Bria This gets very annoying because very often people are put on hold longer than 30 seconds (the phone's default.) In a company with more than 100 soft phones

Hello! I want to get rtcp stats to cdr. (btw, I run asterisk 1.6.2.11) There is howto here http://www.voip-info.org/wiki/view/Asterisk+RTCP But I (and my users) do bridged calls from dahdi to sip, so in h extension channel is dahdi , and it doesn't contain rtcp stats. There is info about function shared. But I can't understand how to use it to put stats from sip channel to dahdi

[Apr 15 11:12:19] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:37259, rtcp halted Operation not permitted [Apr 15 11:12:23] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:38563, rtcp halted Operation not permitted What is the specific nature of this traffic? Despite the above the call still functions. What

Hi, I am plying with * for 10 days now. I am testing with a couple of vocaltec h.323 gateways (FXO and PRI) cisco ata-186 (configured for SIP) and MSN messenger (SIP). They all seem to interoperate. However I have a problem when * is sending calls to the vocaltec gateways. Vocaltec gateways are monitoring the RTCP reports send from the remote gateway (in this case *) and if they don't get a

I have a private network like this: +-----------------------+ | firewall | +-----------------------+ | +-----------------------+ | 1.2.3.4 |