Tag : sip
Home » Posts tagged sip" (paged 12)
Im looking at setting type=peer vs type=user (in both IAX and SIP conf entries), and I found a comment attributed to digium (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that type=user is depricated and that we should only ..
everyone.Having experimented a but with a prototype of a system I described in an earlier thread (Reading DTMF sent by callee during a SIP call), Idecided to implement my requirement by transferring the call to another extension. This way, the cal..
My target system is :PSTNSip ProviderRouter with fw/NATAsteriskSIP PhonesAsterisk is configured to keep NAT connection with SIP provider open (with qualifyfreq) so I dont have any problem (yet) with either casual incoming or outgoing calls.To work aro..
The Asterisk Development Team is pleased to announce the second beta release of Asterisk 12.0.0. You can immediately download this release at http://downloads.asterisk.org/pub/telephony/asterisk/releasesWe strongly encourage all interested Aster..
Follow these steps to configure OpenVPN + SIP : 1. Install OpenVPN on Asterisk server. On appliance, there’s only a single binary /bin/openvpn, and configuration files are in /et..
To put it simply, is the process where Asterisk tries to redirect the RTP media stream to go directly from the caller to the callee. Be careful that some devices do not support this (especially if one of them is behind a NAT). The default setting..
Decrementing the CSeq header field value is directly in violation of RFC 3261. From section 22.2: When a UAC resubmits a request with its credentials after receiving a 401 (Unauthorized) or 407 (Proxy Authentication Required) response, it M..
If after installing Asterisk 10.X.X you see the following errors on SIP reload: No valid Transports Available, Falling Back To UDP, This might be because a possible regression. Please open an issue in JIRA referencing th..
The Asterisk Development Team has announced the release of Asterisk 1.6.2.22. This release is available for immediate download athttp://downloads.asterisk.org/pub/telephony/asterisk/The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sam..
Asterisk Project Security Advisory – AST-2011-014Summary: Remote crash possibility with SIP and the automonDescription: When the automon feature is enabled in features.conf, it is possible to send a sequence of SIP requests that cause Aster..