similar to: Weird LAN VoIP Echo

I am having a problem with Red Alarms on X100P cards. The most frustrating thing is I can not duplicate the alarms, therefore am not sure how to solve it. I have read after searching posts and the web that you can try different signaling methods which may help alleviate the problem. There is fxsks (Which I am currently using), fxsgs, and fxsls While reading the Digium site about these different

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, We have 15 Soundpoint IP 500 phones and recently call quality has deteriorated. On some calls there is a static-buzzing of sorts that occurs when users talk. It can be picked up on SIP-SIP calls and SIP-ZAPTEL (Channel BANK<<-T100P). It basically sounds really weird whenever someone talks, it sounds like a bee buzzing or something. Very hard to explain. Also, there will be echo

Check with your telco. We had the same problem on 1 of our PRI's, every day at 5:00 sharp, red alarm, with all calls cut off for 30 seconds exactly. Turns out the equipment at the CO was going into a test loop at that time because of a forgotten setting by a tech. Man, what a finger pointing exercise that was. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent:

Hello, About every 2 or 3 days I notice in the messages log file: Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 2: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 3: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 4: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 5: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on

Hello, After reading the lists and taking reccomendations from TC, I have finally given up on the echo can built into asterisk. I am sick of hearing complaints from users, so the money spent on a hardware echo can will be worth its weight in gold. I am curious however, about some setup and component requirements. It seems as if every telecom place I call, either never calls back or doesn't

Hi Lars, Thanks for the help ** Test 1 ** On VPN_office I use 'tcpdump -npi any icmp and host 192.168.1.3' When pinging 192.168.1.1 from client 1, with no success, I see no packet passing. When pinging 192.168.1.3 from client 1, with success, I see packet passing: IP 172.16.0.3 > 192.168.1.3: ICMP echo request, id 2648, seq 23, length 64 IP 192.168.1.3 > 172.16.0.3: ICMP echo

Le mar. 15 janv. 2019 à 03:13, Lars Kruse <lists at sumpfralle.de> a écrit : > Hello Julien, > > > Am Mon, 14 Jan 2019 22:15:47 +0100 > schrieb Julien dupont <marcelvierzon at gmail.com>: > > > > ** Test 1 ** > > On VPN_office I use 'tcpdump -npi any icmp and host 192.168.1.3' > > When pinging 192.168.1.1 from client 1, with no success, I

Anyone interested in a lot of gently used IP500's with SIP Image? Also, anyone in need of large quantities of SPA-2000's at an extreme discount? If so, please contact me off list. Thanks, Garrett Smith gsmith@voipsupply.com B2 Technologies/ VoIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM:

Keith, Thanks for the reply and the pointers. > Did you remember to activate kernel ip forwarding? > i.e. echo 1 > /proc/sys/net/ipv4/ip_forward ? I actually forgot to do this, but I have enabled it now in /etc/systctl.conf and can confirm now after a reboot that it's enabled. Unfortunately, still can't ping the node on the LAN. > and when I saw that I was about to cancel

Hello Julien, Am Mon, 14 Jan 2019 22:15:47 +0100 schrieb Julien dupont <marcelvierzon at gmail.com>: > ** Test 1 ** > On VPN_office I use 'tcpdump -npi any icmp and host 192.168.1.3' > When pinging 192.168.1.1 from client 1, with no success, I see no packet > passing. Sorry - the tcpdump command should end with "192.168.1.1" instead of

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

The game runs really well, high fps etc...But when I either enter combat, or sheath/unsheath my weapon, the game locks up for a few seconds ;( Anyone ever have this problem, or have any clue as to where it's coming from? My computer specs are P4 3.0ghz ATI Radeon 9800 pro 128mb 768mb ddr ram Thanks in advance.

Hello, I''m using Linux kernel 2.6.x and tc (from iproute2 package). I''m trying to use HTB or HFSC scheduler in order to limite the rate of outgoing packets and also in order to minimiez delay for RTP stream. But I didn''t suceed in having this 2 QoS services working. I use Iptables in order to classify packets. Here is my HFSC conf. In fact the pings that i send from

We have Polycom IP500's, and just starting recently (after the broadvoice patch I might add) after about 1-2 days these phones ring, and answer, but we get no audio on the phones. The caller can hear us, but we cannot hear the caller. Its happened 4-5 times and is only intermittent. No errors on the console, using g.711u. Any ideas? Tim Jackson Network Engineer Angelina County, Texas

Hello all, I've got myself into a bizzare situation that I can't seem to get myself out of... Was wondering if anyone had some advice that might get me 'over the hill' on this... Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco 7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are all on a separate LAN. There is

Hey everyone The question that i am putting up will sound a bit odd as i am a newbie to asterisk. I have downloaded and install the asterisk and had a look at some of the configuration files like sip.conf, users.conf and extensions.conf. Now my question is that I want to do voip with another pc in LAN, what shall i have to do to implement this and i guess it will be possible without any hardware

Aloha, Does anyone know of any opensource or inexpensive software that can test a network for a possible VoIP installation? I have seen Chariot by NetIQ, but it is very costly! I am envisioning two or three "client" devices that talk to a server device. Each client could simulate 1-100 VoIP phone calls. The server module would monitor the QOS, jitter, etc., and provide some kind of

i have a customer that needs to plug the phones into the pc's using the pass-through rj45 available on most sip phones the question they are asking me is how to keep the data network separate from / secure from the voip network i understand they can set up vlans but i am hazy on a few details 1 since the phones are plugged into the pc's how will the phones be segmented into their own

I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if