similar to: DSL Provider

Hello, are the gsm codecs in Cisco as53xx or 36xx (gsm, gsmfr) copatible with Asterisk's gsm codec? Thanks in advance, Thomas

Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a

Colin, Thanks for your assistance. Reading over your advice I seem to still be a bit confused. My agents are not on the Asterisk server; it appears in your advice that my the call will travel this path: WWW interface --> agent enters their DID, platform to use, and termination DID --> AST calls agent --> Agent calls termination DID If my agents are not on the Asterisk server

Hello, I'm not sure what you mean, could you elaborate? Thanks, -- -- -- Christopher T. Aloi USA Datanet - Technical Support Engineer 318 South Clinton Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074 E: caloi@usadatanet.com <mailto:caloi@usadatanet.com> -- -- -- _____ From: Wojciech Tryc [mailto:Wojciech.Tryc@pikatech.com] Sent: Friday,

Hi ups-devel, As I looked throw www.networkupstools.org site, haven't found any pointers where to send patches, so I hope this is the right place. (Please let me know weather this is the right place.) The patch contains a minor change in the logic of the driver, our Effekta MHD 3000 UPS continuously writes status data to the serial port, so one has to send a CR character and empty the

People, My goal is to get asterisk dialing out via my landline (POTS) from a sip softphone. Ive got the phone, The TDM400p is installed and working. (See below) When ever I dial a number that is directed to the outgoing port on my card (fxs/fxo?) I get no ringing, then it claims its been answered. the CLI reports the following: Executing Dial("SIP/301-f97a",