search for: senykoff

Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on

...| | | -------------- | LAN Switch | -------------- | | | | Any recommendations on something that isn't as pricey as Cisco? I'm in discussions about us building Linux units down the line, but for now we need something we can buy. Cisco is too expensive for us. TIA! Ron Senykoff Systems Architect / Developer HarrisLogic Inc. 972-215-0488 x 3020 312-404-8745 (cell)

Hello all. I created a "plug and play" kind of solution for bridge-based QoS LAN-WAN traffic shaping. Based on the LEAF bering-uClibc branch, I''m calling it ''QBox'' and the project site for now is located at: http://content.cs.luc.edu/projects/comp412/q-box I''m interested in any feedback you might have. This is based off of experience I have over the

...www.burnpc.com/website.nsf/all/3a64a6369757819686256f960068ad75!OpenDocument If anyone has any questions or feedback I'm very interested. This configuration has evolved over the past 6 months and is pretty tweaked. The highest priority traffic in my config, is of course, IAX2. Regards, Ron Senykoff Systems Architect / Developer HarrisLogic Inc. 972-215-0488 x 3020 312-404-8745 (cell) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050127/46e600ce/attachment.htm

I have the following situation: Asterisk 1.2.1 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 Some 501's local to my network, some across the great INTERNET divide. PRI connected to Sangoma card. Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker phone when calling from extension to extension. echo not present when calling outbound

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

I have a very simple setup exactly as described in the HOWTO section " 4.2. Routing for multiple uplinks/providers". One is cable (eth1: dhcp) and the other is PPPoE (ppp0). I used the following commands to configure the routing once all of my interfaces are up and i have configured SNATing for them: ip route add 66.11.173.0/24 dev ppp0 src 66.11.173.224 table 11 ip route add default

...t 1:0 prio 1 u32 match ip dport 5036 0xffff flowid 1:5 # IAX2 # tc filter add dev $e protocol ip parent 1:0 prio 1 u32 match ip sport 4569 0xffff flowid 1:5 tc filter add dev $e protocol ip parent 1:0 prio 1 u32 match ip dport 4569 0xffff flowid 1:5 #### Remotely VPNd SIP Phones ##### # Ron Senykoff Home tc filter add dev $e protocol ip parent 1:0 prio 1 u32 match ip src XXX.XXX.XXX.XXX/32 flowid 1:5 tc filter add dev $e protocol ip parent 1:0 prio 1 u32 match ip dst XXX.XXX.XXX.XXX/32 flowid 1:5 ##################################### ############### Video ############### #...