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Using either Asterisk or FreeSwitch and libraries such as Skypiax, I need a server that upon creation...creation of a new Skype (i.e. skype username and password) user in MySQL DB, signs this user into the server...server and enables me to send SIP traffic with URI something like "sip:fromskypeuser-toskypeuser at sipskypeserver
Complete training of Asterisk and A2billing along with the extensions , dial plan, and other related...configuring asterisk , basic trouble shooting. Installation of A2Billing and compiling with Asterisk. Configuration...in Voice Chat VIA SKYPE.
The person should also assist in Finding the cheaper SIP trunk and DID provider
...for routing sip calls to Skype. We need an Skype API written so we call route incoming sip calls to any...any Skype user account. We would like to use asterisk real time or any other mythod that would allow us
my sip provider only allow 1 outgoing sip call per account. i plan to register 5 or more new accounts...is bascially some kind of SIP trunk that combines all 5 accounts (SIP1 - SIP 5) in one trunk.
_Call flow...caller calls, asterisk will pick up SIP1 and dial
when a second person calls, and asterisk will use SIP2
...configure or program our Asterisk (using Free PBX) so when SIP calls comes to Asterisk, the call immediately...immediately route to Skype network without checking the extension or DID on the database. We want to avoid the...the database checking in the Asterisk and send directly to Skype (we already have Digium license).
i am trying to connect to a voice provider using SIP. all i am given is a username, password and Ip address...problem there. however, when i try to dial out using asterisk. i got stuck there.
sip.conf...conf [general] bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = xx.xx.xx.xx ; Address to bind
...the network requirements:
Calls from PSTN -> to SIP gateway -> ask for destination -> connect to IP Phone_or_Softphone...with a DID 1-555-xxx-xxxx)
Calls from Skype -> Skype to SIP gateway -> ask for destination -> connect...connect to IP Phone_or_Softphone
(skype to sip is either hardware or software)
Call from IP phone -> ask
...We have a Linux server running Asterisk.
It's running a Multitenanted PBX system with SIP VoIP accounts...registered to a remote SIP server.
Occasionaly (for some unknown reason) all of the SIP acocunts become unregistered...until we reboot the box.
Your Linux code needs to check the Asterisk registration log on a regular basis
The Rhino SIP-IAX2 Proxy is a software product that interfaces with Rhino's PCI and PCI Express cards...“last mile.?? The software will be available for Linux, Windows, and Mac OS X and will allow a connection...industry-standard protocols, specifically:
• SIP (RFC 3261)
• IAX2 (RFC draft-guy-iax-04)
...voip call termination
server A is the main asterisk ,server b,c,d.....will multiple server for end...end node
the system will perform as sip call--->serverA-------->iax2-------->server(server a)--...the prefix of the gateway
pls kncok me on skype iptelbd
Have a linux server on Amazon Web Cloud, need:
1. asterisk installed and configured
2. install and...and configure webmin
3. connect to sip termination provider
4. setup first couple extensions and voicemail
A Asterisk is installed and a script is created that calls a SIP. The script is triggered from a Pyton...workflow and the soundfiles was setup on another Linux server and worked had OK sound quality
The...is to help correcting the soundquality on the Asterisk/SIP connection
I have Asterisk 1.8 installed on CentOS
I have manged to config the sip and extension files to...to connect to the SIP Provider (Test account) and it's working fine.
You will need to teach me how...how to setup multiple operators on Asterisk, for example - having 10 phones...
Also you will need
I would like to use asterisk as a softphone with Vtiger in conjunction with a french voip gateway.
The...The job: Asterisk installation script and configuration depending of the chosen french voip gateway....coder's Seller Legal Agreement).
I have done my asterisk training need recap and training full again. I need training on following:
4) Installing Asterisk server and learning about Everything extensions, sip trunks , Inbound and...etc etc full
5) Using MYSQL for call history and sip users and international calling etc etc.
...need help in configure my asterisk box to connect to Huawei SoftX3000 SIP server. i managed to registered...ast_generic_bridge: Didn't get a frame from channel: SIP/xxxxx-082e96b0
Aug 31 15:23:53 DEBUG: channel...ast_channel_bridge: Bridge stops bridging channels SIP/xxxx-b7a05ad8 and SIP/xxxx-082e96b0