search for: sysmaster

Hello Guys, After adding money into my sysmaster phone account I am able to make calls outside.thnx _____ From: Mani Nair [mailto:mnair at nvloisp.com] Sent: Friday, September 07, 2007 9:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Sysmaster and Asterisk Hello Guys, I am unable to make cal...

...isk acts as Inbound Gateway and a. Sends Out of-Network PSTN Calls to SIP enabled PSTN Provider b. Sends all DID calls to Asterisk Users using SIP protocol 6. Asterisk handles the both In-network and Out of Network calls 3rd Scenario: 7. Asterisk will be registered as SIP endpoint (Outbound) in SysMaster gateway 8. VOIP Customer uses H.323 Broadband endpoint and calls into SysMaster GK 9. SysMaster Sends the DID Calls to Asterisk as SIP Traffic, using it's SIP Interface. All other H.323 calls will be routed to H.323 Terminators/Gateways 10. Asterisk acts as Outbound Gateway for calls to Registe...

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

Is there anyone with experience how to integrate Sysmaster's VoiceMaster? Please can you share your experience. Thanks. Habiyakare Aimable Voice Services Terracom Communications Tel :(250)08435550 SIP:04400104@voice.terracom.rw E-mail:aimable@terracom.rw MSN:aimable@terracom.rw -------------- next part -------------- An HTML attach...

...XX.69:5060 SIP/2.0 Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c Max-Forwards: 70 From: <sip:321445xxx6 at 80.XX.XX.69>;tag=f395877e02bf8eb2fd8f5a0e To: <sip:329298yyy6 at 80.XX.XX.69> Call-ID: f395877e02187250fd8f5a0f at 80.XX.XX.68 CSeq: 1 INVITE User-Agent: SysMaster VoIP Gateway v1.2.0 Contact: <sip:321445xxx6 at 80.XX.XX.68:5060> Remote-Party-ID: <sip:321445xxx6 at 80.XX.XX.69>;party=calling;screen=yes;privacy=off Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE Content-Type: application/sdp Content-Length: 261 This is my sip.conf : [ou...

...Member(). Right now, even though Samba is configured OK they always return false, even if a user is member of a group. I can use User Manager to view user and group information... Am I trying to use functions not implemented yet? Thanks, umberto -- Umberto Nicoletti - unicoletti@arpa.veneto.it | sysmaster@arpa.veneto.it Tel. 049-8239380 (assistenza) "We'll try to make different mistakes this time." - Larry Wall

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients...... plz anyone suggest me.... Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061227/aa4e409c/attachment.htm

Sherwood, Your intentions are noble and your desire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com)