[asterisk-users] asterisk SIP MESSAGE method support
Hello List, I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call. But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). Is it possible to do this in asterisk using some tricks? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.comhttp://www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk SIP MESSAGE method support
2011/4/7 Deka, Rajib IN MAA SL rajib.d...@siemens.com Hello List, I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call. But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). There is ongoing development to enhance Text support in Asterisk's trunk. Out-of-call messaging is one those features. Regards Is it possible to do this in asterisk using some tricks? Regards, *Rajib Deka* SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com -- Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk SIP MESSAGE method support
Is the following is the link for getting the source, http://svn.asterisk.org/svn/asterisk/trunk/ Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com Sent: Thursday, April 07, 2011 6:20 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 81, Issue 19 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. asterisk SIP MESSAGE method support (Deka, Rajib IN MAA SL) 2. Re: Iptables configuration to handle brute force registrations? (Gilles) 3. Re: BRI Configuration help me (mahesh katta) 4. Re: Iptables configuration to handle brute, force registrations? (Gilles) 5. Compiling asterisk using NDK build (Nikhil) 6. Re: asterisk SIP MESSAGE method support (Olivier) 7. Re: BRI Configuration help me (Tzafrir Cohen) 8. Re: Compiling asterisk using NDK build (Tzafrir Cohen) 9. Re: BRI Configuration help me (mahesh katta) 10. Re: Trunk form asterisk1 to asterisk2 fails (GiGi) 11. Re: Asterisk 1.8.3 (Satish Patel) 12. Re: Asterisk 1.8.3 (Bryant Zimmerman) 13. Re: BRI Configuration help me (mahesh katta) -- Message: 1 Date: Thu, 7 Apr 2011 14:54:23 +0530 From: Deka, Rajib IN MAA SL rajib.d...@siemens.com Subject: [asterisk-users] asterisk SIP MESSAGE method support To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Message-ID: 2658e54b540d284981ea57e6a549ea70a592f02...@inblrk77m1msx.in002.siemens.net Content-Type: text/plain; charset=us-ascii Hello List, I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call. But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). Is it possible to do this in asterisk using some tricks? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.comhttp://www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/8ec3b210/attachment-0001.htm -- Message: 2 Date: Thu, 07 Apr 2011 12:51:48 +0200 From: Gilles codecompl...@free.fr Subject: Re: [asterisk-users] Iptables configuration to handle brute force registrations? To: asterisk-users@lists.digium.com Message-ID: ko5rp6huuoqu2suivok9f0p0nccb4n9...@4ax.com Content-Type: text/plain; charset=us-ascii On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: Have a look at these: Thanks much Gordon. I'll study the scripts you mentionned. It looks like iptables is good enough and I won't have to install a second tool to watch the logs and reconfigure iptables on the fly. -- Message: 3 Date: Thu, 7 Apr 2011 16:48:13 +0530 From: mahesh katta maheshka...@flexydial.com Subject: Re: [asterisk-users] BRI Configuration help me To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: BANLkTikP-CfWjOGw5--D48EuHT=afr_...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Sir, my files are in fistmail that is my configuration. and till its disconnecting the line On Thu, Apr 7, 2011 at 2:35 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: Hi, Un-top-posting On Thu, Apr 07, 2011 at 12:53:18PM +0530, mahesh katta wrote: On Wed, Apr 6, 2011 at 9:04 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Wed, Apr 06, 2011 at 07:15:00PM +0530, mahesh katta wrote: Sir, i am using goautodial server , bri card is showing ok but when i try to call
Re: [asterisk-users] asterisk SIP MESSAGE method support
On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote: Is the following is the link for getting the source, http://svn.asterisk.org/svn/asterisk/trunk/ Please try not to reply to the entire digest.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk SIP MESSAGE method support
Is the following trunk has development version of out-of-call messaging capability, also what is the version of asterisk, http://svn.asterisk.org/svn/asterisk/trunk/ Regards, Rajib -- Message: 10 Date: Thu, 7 Apr 2011 14:42:35 +0100 From: Steven Howes steve-li...@geekinter.net Subject: Re: [asterisk-users] asterisk SIP MESSAGE method support To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: d5d50321-4b5b-41bd-b8a3-8bcceafc2...@geekinter.net Content-Type: text/plain; charset=us-ascii On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote: Is the following is the link for getting the source, http://svn.asterisk.org/svn/asterisk/trunk/ Please try not to reply to the entire digest.. S Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk SIP MESSAGE method support
On 11-04-07 09:59 AM, Deka, Rajib IN MAA SL wrote: Is the following trunk has development version of out-of-call messaging capability, also what is the version of asterisk, http://svn.asterisk.org/svn/asterisk/trunk/ I don't believe the branches has been merged into trunk, you can use russellb's branch [1]. [1] http://svn.digium.com/svn/asterisk/team/russell/messaging/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
