similar to: Help ! Echo on sip call.

I have a * server that receive calls from the internet interface (eth0, external) and from the local interface (eth1, internal), when the calls are between sip phones from the local interface (internal) all is ok, no echo, but when the call is between a external sip phone and a internal sip phone there are echo, a delay of 500ms, only on the internal side, the external side don't hear the

-----Mensagem original----- De: miguel@amplanet.com.br [mailto:miguel@amplanet.com.br] Enviada em: quarta-feira, 11 de agosto de 2004 16:31 Para: 'asterisk-users-admin@lists.digium.com' Assunto: Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted (tethereal) Capturing on eth1 2.738748 192.168.199.4 -> 192.168.199.5 SIP/SDP Request: INVITE

Hi. I just went thru reading all of the howto and have a working implementation of RSVP over UDP encapsulation that i would linke to test, made in Java. For this, I would like to use a Linux Box as a RSVP enabled router. However, I have tried and tried to make RSVP work on linux and failed. So I wonder what I am doing wrong. Basically, on the linux box what I am doing is turning

Hello again, First, excuse me for my poor english. I''m trying now to make bandwith control in a firewall machine running Shorewall. This machine is also a bridge using bridge-utils bridge-utils-devel. It is a mandrake 10. The configuration is something like this: FTP/Webserver ------| eth0 eth1 Mailserver

I have Asterisk running on my LAN with softphone clients (SJPhone) and Cisco 7940/60s, all using SIP. I also have a few remote sites connecting to my Asterisk server. I am getting an echo back of my voice when talking with one particular site. The caller does not hear an echo on their end. All calls on the LAN or to other sites do not produce an echo. When the caller places his SJPhone on

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I have a weird (?) problem here on a setup running CentOS 5.3 x86_64 (and OpenVZ, and some home-brew L2TP daemons, RIPd, BGPd, etc). There's a (VE in OpenVZ speak) virtual machine that has two ethernet interfaces, seen as eth0 and eth1, respectively. Those live in VLANs, but it's not important here. The thing is that on eth1 the

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

Hi all ! I''m building a network appliance whose goals are to enhance Voip quality on 512/128k DSL links. But, i have a voice quality problem, and i think i''m doing something wrong, but i can''t find what. Please excuse my terrible english, i''m french. >From what i''ve read, tc is the tool i need. I read the documentations, readmes and lartc.org

In my hunt to track down my echo issues, I tried disabling all echo cancellation, suppression, adaption, on my SPA-2000 (Advanced section of the config, under Line 1/2). Then calling from one local extension to another. (SPA-2000 Line1, to Line2 on the same device) I was pretty shocked with the results, the echo was HORRIBLE! I even tried 3 different analog phones. Now, once I turned the echo

Hi Guys, I have a couple of problems that are driving me nuts. This is the script I am using to setup restirctions and bw limitations: #Adding root class. tc qdisc add dev eth0 root handle 1: htb default 10 &>/dev/null tc class add dev eth0 parent 1: classid 1:1 htb rate 100Mbit ceil 100Mbit burst 2k quantum 1536 tc qdisc add dev eth1 root handle 1: htb default 10 &>/dev/null tc

I have a problem: When I'm in a call and a second call arrive (call waiting) I can't transfer the first call. If I press flash the line change to the second call, if I press flash again the line change to the first call. How I can transfer a call in this kind of situation ? Kind regards, Miguel

hi all, my system with kernel 2.4.16 & 2.4.17( RH-7.2) has netfilter-1.2.4, iproute-ss010803, squid, dns with 4 ether cards.i am trying a simple situation where i want to shape traffic all the interfaces. at present iam using netfilter to allow a few ips, using transparent squid & iproute to limit bandwidth. i feel i must give more bandwidth for my local servers for my office &

Hi 4 all, Best regards from Lihtuania, I have one question: I'm runing slackware linux 4.0 wiht kernel 2.2.12, using samba 2.0.5a I have 2 ethernet card's and 1 wavelan card. eth0 - wavelan. ifconfig eth0 192.0.0.3 netmask 255.255.255.224 broadcast 192.0.0.31 (wavelan) ............ ifocnfig eth1 192.0.0.65 netmask 255.555.255.224 broadcast 192.0.0.95 (samba client's on this

I have 2 machines running centos. each machine has 2 network cards. Box 1: eth1: T1 connection 74.X.X.X eth0: office connection 192.X.X.X Box 2: eth1: Cable modem 24.X.X.X eth0: office connection 192.X.X.X How do I issue a route command on each box so it will use eth0 as the shortest route between the two boxes??? Thanks, Jerry -------------------- On Box1 the route is: [root at mndemo ~]#

Hello again, First, excuse me for my poor english. I''m trying now to make bandwith control in a firewall machine running Shorewall. This machine is also a bridge using bridge-utils bridge-utils-devel. It is a mandrake 10. The configuration is something like this: FTP/Webserver ------| eth0 eth1 Mailserver

Hi People, what files is processed first?, balcklist or rules, i want to globally filter imesh, but at the same time allow managers to connect, i.e. , imesh work on port 1214, i have this: /etc/shorewall/blacklist #ADDRESS/SUBNET PROTOCOL PORT 192.168.0.0/16 tcp 1214 192.168.0.0/16 udp 1214

-------------- eth0------eth1 eth0------------ |211.241.219.xx | --- | ROUTER | --- |192.168.1.4 | --------------- --------- ------------ when i send traffic from ROUTER to 211.241.219.xx or 192.168.1.4(masquraded), the filter works fine... In ROUTER, tc filter policy is like this: tc filter add dev eth0 parent 1:0 protocol ip u32 match ip dport 80 0xffff

I'm a problem with bridge please help me My configuration: eth0,eth1,wlan0 bridge is br0 on devices eth0 and wlan0. Device eth1 is not in bridge. br0 ip is 192.168.1.1 eth1 ip is 192.168.2.1 ethernet (eth0,eth1) is in one switch. all is ok if eth1 not in system, if eth1 plugged into system and up interfa= ce eth1 network crash :( thanx. -------------- next part -------------- An HTML

Hi all. Just got my 'Developer Kit Lite', installed it, and made the changes to load the modules in kernel and in the configuration files. Call thru-from fxo and the fxs sound great. Even fxs-iax-sip sound ok. When a answer a call coming into asterisk from the PSTN thru the fxo i have a loud echo. I have echocancel=yes. Is there another parameter I can change in the

On Tue, Nov 8, 2016 at 8:02 AM, Boris Epstein <borepstein at gmail.com> wrote: > Hello there, > > What is the hypervisor that hosts the VM? What does ifconfig show on it? > > Boris. > > On Mon, Nov 7, 2016 at 4:36 PM, Gordon Messmer <gordon.messmer at gmail.com> > wrote: > > > On 11/06/2016 11:00 PM, Sean Son wrote: > > > >> How do I