search for: sipconnect

Has anyone had any experience connecting Asterisk to Cbeyond's SIPconnect service (http://www.sipconnect.info)? Any opinions? Thanks, -Brian

...4,10000-20000 to asterisk server Firewall on asterisk server is switched off (for better testing) [sip.conf] [general] port=5060 bindaddr=0.0.0.0 language = de externhost=intakt-musik.dyndns.org externip=intakt-musik.dyndns.org localnet=192.168.0.0/255.255.255.0 nat=yes register => USER:PASS at sipconnect.sipgate.de/USER [sipconnect.sipgate.de] type = friend host = sipconnect.sipgate.de outboundproxy = sipconnect.live.sipgate.de port = 5060 username = USER fromuser = USER fromdomain = sipconnect.sipgate.de secret = PASS dtmfmode = rfc2833 insecure = port,invite ;insecure = very canreinvite = no reg...

...r SIP phones is registered with the Trixbox and are set to use the siptrunk for outbound calls. So, outbound calls (SIP/Asterisk -> PSTN/Broadworks) work properly. The RPID is set to the TGID and the From is set to the CLID from the SIP phone user. Broadworks likes this, treats these calls as sipconnect 1.0 type 2 calls, and the outbound calls are successful. However, inbound calls are signaled similarly with RPID and To, but the Asterisk keys on the RPID. We have no user or route for the TG number and therefore the calls fail. We don't want to route based on the RPID, but on the To as the...

...e same conference - have a nice chat - play an announcement and kill the conference after a max time value so far so good. I developed the following Dialplan to fulfill the task (simplified, hopefully right - deleted all announcements, cdr, etc. :)): [macro-c2cmacro-p1] exten => s,n,Dial(SIP/sipconnect/${to1},120,M(c2cmacro-connect^${ARG1}^to1)) exten => s,n(lbl_macro-c2cmacro-p1_1),MacroExit() exten => s,n(lbl_macro-c2cmacro-p1_2),Hangup() exten => h,1,MeetMeAdmin(8000,K) exten => h,n,Goto(s,lbl_macro-c2cmacro-p1_1) [macro-c2cmacro-p2] exten => s,n,Dial(SIP/sipconnect/${to2},120...

...? Geolocation: <cid:target123 at atlanta.example.com> Geolocation-Routing: no 2. Reciprocally, how can you read such application/pidf+xml section an incoming call ? 3. What do you know of this RFC 6442 adoption within SIP industry ? Best regards [1] https://www.sipforum.org/download/sipconnect-technical-recommendation-version-2-0/?wpdmdl=2818 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20171207/435711f2/attachment.html>

Ever since upgrading to 1.6 I get messages like these. I want everything else that shows up, but is there a way to make all the dns messages go away? Ira > doing dnsmgr_lookup for 'gw5.telasip.com' > doing dnsmgr_lookup for 'sipconnect.ipcomms.net' > doing dnsmgr_lookup for 'proxy.ideasip.com' > ast_get_srv: SRV lookup for '_sip._UDP.proxy.ideasip.com' mapped to host proxy.ideasip.com, port 5060

Hi, I am just about to move out from my parents home and think about how I will phone from now on. In Germany there is a provider (QSC) who offers DSL (1024 down/256 up) with fastpath without volume or time limits. Does anybody know a comercial (or even semi-professional) provider who lets me dial out through H323 (or another protocol) and also offers an number where I can be called from

Hello, When I get a SIP INVITE as follows: INVITE sip:s at 10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: "0475XXXXXX" <sip:1053212 at sip.domain.com>;tag=as7df9ab18 To: <sip:02XXXXXX at IP:5060> Contact: <sip:1053212 at IP:5060> Call-ID: 344d42bd16975a54141d11f635bdfc71 at sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL,