similar to: sip show channelstats shows all 0

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 ( %)

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

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:

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

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 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

Hello! I'm just wondering what the RTT exactly means. Where are the exact measuring points located? > pjsip show channelstats ...........Receive......... .........Transmit.......... BridgeId ChannelId ........ UpTime.. Codec. Count Lost Pct Jitter Count Lost Pct Jitter RTT....

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

Google says Round Trip Time https://www.voip-info.org/asterisk-rtcp/ Doug

On Sat, May 16, 2020 at 10:58 AM Michael Maier <m1278468 at mailbox.org> wrote: > On 15.05.20 at 14:31 Doug Lytle wrote: > > Google says Round Trip Time > > > > https://www.voip-info.org/asterisk-rtcp/ > > That doesn't answer my question (I know the abbreviation RTT). Therefore > I'm trying again: > > I'm just wondering what the RTT *exactly*

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,

Hi people, the following bug makes some error messages to appear al log level 0 when they are supposed to be al level 3. Any guy at the printer team could tell if I am safe printing with all those messages and why they are generated? Only a few computers are affected in my network. https://bugzilla.samba.org/show_bug.cgi?id=10118 Thanks! On Oct 25, 2013 1:21 PM, "Ezequiel Larrarte"

> Asterisk Project Security Advisory - ASA-2007-011 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Multiple problems in SIP channel parser

> Asterisk Project Security Advisory - ASA-2007-011 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Multiple problems in SIP channel parser

On 17.05.20 at 01:28 Joshua C. Colp wrote: > On Sat, May 16, 2020 at 10:58 AM Michael Maier <m1278468 at mailbox.org> wrote: > >> => How are the RTT values exactly calculated? Which values are actually >> used for? >> > > The value is calculated according to the logic in the RFC[1]. Specifically > using embedded timestamps in the RTCP packets and

Hi, I recently decided to change my setup from AsteriskNow to plain-asterisk 1.4, which I wanted to set up and configure myself on a server running Debian Etch 64bit version. Hardware: Asrock motherboard, model 775Dual880-Pro, with a Celeron D running at 2.8GHz, 1GB memory, standard Nvidia GF4MX videocard, and one X100P clone card. Running the AsteriskNow, everything worked fine, except for

On 15.05.20 at 14:31 Doug Lytle wrote: > Google says Round Trip Time > > https://www.voip-info.org/asterisk-rtcp/ That doesn't answer my question (I know the abbreviation RTT). Therefore I'm trying again: I'm just wondering what the RTT *exactly* means. Where are the exact measuring points located? => How are the RTT values exactly calculated? Which values are actually

Hello, I've been searching around the net all day today and i can't seem to find much info that's helping with a few issues i've been having. Background: using AsteriskNOW beta5 (asterisk 1.4.2) with mysql real time configuration, Currenlty only have 4 sip users setup and 1 queue. When i call into the queue upon connecting to the agent (ie it gets past the IVR stuff) i recieve the

Hi Everyone. I'm in a quandry & don't know which way to go. - Obviously I'm an Asterisk newbie although I've been watching this list for over 2 years now. I've got an Asterisk box (actually, it's an AsteriskNOW box) up and running here at home. - It's on my home LAN - NAT'ed behind my LinkSys router. - On the same LAN I've got a Cisco 7940, 7960, and

Hi, there. I have asterisknow beta 5 with the following data: Ip 192.168.0.60 mask 255.255.255.0 gw 192.168.0.1 the router (a linksys) has port forwarded the port udp 5060 and from 16384 to 16482 udp-tcp from the internet to the asterisk machine. the only protocol allowed is g729. Which work fine for the ip phones I already have setup in the LAN. My problem is trying to register to a voip