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...e a requirement to dialout some external number, once the call is answered the same has to be forwarded to an Internal Queue. Please help me. I have tried calling with two SIP end point forwarding , even that is not working, My dial plan line is , Dial(SIP/19201/19202,300) -- *Best regards,* *Ruban.S* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150903/2c9f09a3/attachment.html>

...Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03 9:09 GMT-03:00 Kantharuban Ruban <kanth.ruban at gmail.com>: > >> Hello Group, >> >> I have a requirement to dialout some external number, once >> the call is answered the same has to be forwarded to an Internal Queue. >> >> Please help me. >> >> I have tri...

...another client (sipml5) it throws the following error: "chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP instance '0x7f3ccc020718'" I am struck here. Please throw some light to go further. Thanks in advance. Best regards, Ruban.S -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150616/21bcf55b/attachment.html>

...ere is the hangup event info, -HANGUP {calleridname=<unknown>, connectedlinename=<unknown>, uniqueid=1384413814.79523, cause=0, cause-txt=Unknown, event=Hangup, privilege=call,all, connectedlinenum=<unknown>, calleridnum=03003, channel=SIP/S0-gateway-0001363d} Best Regards, Ruban.S -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131119/aa07bcbe/attachment.html>

...here, 1) Linux machine is rebooting , we have issued a service asterisk restart command. 2) Tried to connect from an application to asterisk. The manger connections lost. Did anyone faced such issues or any solutions in latest versions of Asterisk to handle such situation. Best regards, Ruban.S Siemens Limited, IC BT SSP ES R&D IPI 1 SP Infocity, Block B, 2nd Floor, #40, MGR Salai, Kandanchavadi, Perungudi, Chennai - 600 096 Tel: +91 44 33524336 Mobile: +91 9500084833 mailto:kantharuban.s at siemens.com -------------- next part -------------- An HTML attachment was scrubbed... U...

...Security Advisory FreeBSD, Inc. Topic: syncache/syncookies denial of service Category: core Module: net Announced: 2002-04-16 Credits: Alan Judge <Alan.Judge@eircom.net> Dima Ruban <dima@FreeBSD.org> Affects: FreeBSD 4.5-RELEASE FreeBSD 4.4-STABLE after 2001-12-14 19:53:01 UTC FreeBSD 4.5-STABLE prior to the correction date Corrected: 2002-02-20 16:48:49 UTC (RELENG_4) 2002-02-21 16:38:39 UTC (RELENG_4_5, 4.5-R...

...going', label 'UP' already in use at priority 33 [Sep 2 16:00:00] WARNING[4970] pbx.c: Extension '_0.', priority 45 in 'outgoing', label 'DOWN' already in use at priority 36 [Sep 2 16:00:01] WARNING[5180] acl.c: Cannot connect Thanks in advance. Best Regards, Ruban.S -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130903/4d74fb05/attachment.htm>

Hello, I`m using typical scheme Samba + Squid + Windows 2003 domain for users authentication, with version 3.0.20.b I have next situation, all authentications request from samb server to AD controller going with samba hostname like example "samba_serv", and I have rule for around 3000 users which use AD only for internet access and not need to use AD for other tasks, rule contain: