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 [Asterisk India | Indian Asterisk User Community Forums - Nurturing the Asterisk talent in India] Not able to make outgoing call [Failed to authenticate on INV ITE]: This SIP server is live server. After registering the two accounts in the SIP server, I can make incoming call from my mobile to the 31045850 but while making outgoing call it is failing and showing

[My Blog] Asterisk Cisco CallManager Voicemail Integration: If a SIP telephone registered to the Asterisk machine acting as voicemail calls through to a Callmanager user and subsequently is sent to voicemail, the call will be dropped. This can be resolved by configuring the device in sip.conf as something like [sipexten] (eg: [sip7222]) rather than [exten] (eg: [7222]).

 [VOIP-info.org Page Changes] chan_dahdi.conf: by h.324video Fri 08 of Aug, 2008We have an extensive investment in home telehealth equipment with a large patient base using pots videophones and store forward devices and need to accommodate patients with bundled VoIP service. Desirable connection speed with pots is 21kbps with v.34 modems.

 [Cyber Tech Help Support Forums] Asterisk / Call Manager 4.2 integration: I was using AsteriskNOW (which has a web interface), and I can't remember if someone else found the line of code for defining the trunk to CM online and set it up, or if it was configured it using a web interface and used a default name. Either way, the trunk link was listed under something like [custom-dialplan1] in extensions.conf, and its name was something like trunk_1.

[vtiger] 3rd-Party Extensions :: RE: Code Contribution: Asterisk CTI ...: - What is wrong : I don't receive any call to my SIP phone. I don't manage to call any phone (SIP neither classic phone).

Reflected tags on Technorati: Blog, Voicemail Conf, VoxxMail