Addpacpekerjaan

Hello, I have addpac cisco GW I want config every one call stop 1200 secs configure terminal voice-port 0/0 timeouts initial 1200 can you config for me

Hello, I have addpac cisco GW I want config every one call stop 1200 secs configure terminal voice-port 0/0 timeouts initial 1200 can you config for me

...SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (GOIP , Addpac , Dinster ,quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) A. Any mini Linux distribution ex...

...calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (GOIP , Addpac , Dinster ,quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) A. Any mini Linux distribution e...

...bandwidth consumption of VOIP termination service provider. By using it, can reduce of bandwidth consumption up to 85% maintaining best voice quality is an innovative technology that not only reduces bandwidth consumption but also work as a strong buffer and a expert coordinator of gateway or all type of VOIP device. Available in CD and flash drive… Compatible with any kind of IP Gateways like Addpac, Dinstar, EuroTech gateways, Quintum analog and digital gateways, GOIP, Cisco, Suncom gateways and any standard sip devices. It will also work with any kind of internet connection like DSL, VSAT, WiMax and 3G by its innovative solution of compressing RTP. Project Description: We need some kind of bandwidth Optimizer system ( up to 80-85 % than usual SIP calls )from S...

...bandwidth consumption of VOIP termination service provider. By using it, can reduce of bandwidth consumption up to 85% maintaining best voice quality is an innovative technology that not only reduces bandwidth consumption but also work as a strong buffer and a expert coordinator of gateway or all type of VOIP device. Available in CD and flash drive… Compatible with any kind of IP Gateways like Addpac, Dinstar, EuroTech gateways, Quintum analog and digital gateways, GOIP, Cisco, Suncom gateways and any standard sip devices. It will also work with any kind of internet connection like DSL, VSAT, WiMax and 2G,3G,4G by its innovative solution of compressing RTP. Project Description: We need some kind of bandwidth Optimizer system ( up to 80-85 % than usual SIP calls ...

i want to access my gateway from other network like on web i m bored to use teamview can you please make online my voip gateway like dinstar,goip,ezoneteach,addpac, also want to use call termination if you can please bid

Hi, If any body can configure my ADDPAC GSM GATEWAY with my Voip Switch directly , please contact me. I want to use it with registration mode , with public IP address. I have no fix or static IP.

We use Csip Simple for one of the Key feature of Filtering and able to rewrite the rules for calls, But facing challenges with its smooth working and Integration like we can using applications Like have a UCS system from addpac korea and we use with Csip now H264 codec & G729 for communcation with desktop phone and between csip to csip Vp8 & with high packet loss or network switching it freeze and don't drop callsn No camera Switch function like castelnqudio quality improvementnWhen negotiating the video with desktop phone seems csip sending delayednnRegardsnVinoj raj

Hi, I am in VOIP business. I want to use my ADDPAC Voip GSM gateways with my Voip Switch directly. If any body can configure it , please contact me.

Project Description: Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to any gateways (quintum /addpac/ dinstar/ eurotec gateway for example) or E1 cards 3. Number of Server B can be unlimited 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) 6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls, billi...

Project Description: Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to any gateways (quintum /addpac/ dinstar/ eurotec gateway for example) or E1 cards 3. Number of Server B can be unlimited 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) 6. For server B need portion of asterisk web gui for adding gateways..prefix or so, viewing active calls...

I do not have the manual company seems to be korean I need all old settings to be removed and new settings to be added namely register 8 sip accounts to a single server and set outgoing dial plan

I have an Addpac 1100F which I would like to use with Asterisk (SIP) I don;t know how to configure the 1100F but I am familiar with asterisk I need for someone to config the 1100F ( you have to use telnet as web interface is not working ) Once done, you can give me the asterisk config and I can do the rest