Re: [asterisk-dev] Placing a new outbound call using C
On Thu, Mar 08, 2007 at 08:48:14PM -0500, Sean Bright wrote: This is only true if he intends to redistribute his code. Right. For example: if he intends to redistribute it to a client. (also remember that a standard unix/linux system includes an atd) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Placing a new outbound call using C
On Friday 09 March 2007 02:45, Steve Edwards wrote: On Thu, 8 Mar 2007, Joel wrote: Compared to calling a function in C there is a significant time difference that is only further amplified by scheduling lots of calls in advance. Anyways, to make it perfectly clear, as the subject states, I want to do this using pure c. Look in manager.c and see how the Originate action places a call via C Paul -- Paul Hewlett Technical Director Global Call Center Solutions Ltd, 2nd Floor, Milnerton Mall Cnr Loxton Koeberg Roads, 7435 Milnerton [EMAIL PROTECTED] www.gccs.co.za Tel: +27 86 111 3433 Fax: +27 86 111 3520 Cel: +27 76 072 7906 Gizmo: 1 747 659 6171 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Re: Placing a new outbound call using C
In article [EMAIL PROTECTED], Joel [EMAIL PROTECTED] wrote: Compared to calling a function in C there is a significant time difference that is only further amplified by scheduling lots of calls in advance. Anyways, to make it perfectly clear, as the subject states, I want to do this using pure c. Joel, I think the part of the picture we are missing when trying to help you is this: you plan to write some functions in C that will directly call the C API in Asterisk; how do you plan to call those functions that you have written? What part of your overall system will cause them to be executed? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Re: redirecting a call after a Queue
DB == Dov Bigio [EMAIL PROTECTED] writes: DB So, the idea would be to have a parameter for the Queue app that DB would be the extension to transfer the call to in case the call is DB finished by the agent. You can catch the call in the h extension (with a Goto). We have been playing with that; unfortunately not with complete success. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] New committer: Marc Blanchet
welcome in da house Marc. ;) On 3/9/07, Olle E Johansson [EMAIL PROTECTED] wrote: Marc Blanchet is a new committer with permission to create branches. He's focused on porting Asterisk to IPv6, something we started discussing the very first Astricon in Atlanta. Marc has been very active in the IPv6 community for a long time and has published books on the topic too. Welcome, Marc! On behalf of the Asterisk Advisory Council /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- Clod Patry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
RE: [asterisk-dev] Placing a new outbound call using C
On Fri, 2007-03-09 at 01:29 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 8 Mar 2007 22:05:42 -0800 (PST) From: Steve Edwards [EMAIL PROTECTED] Subject: RE: [asterisk-dev] Placing a new outbound call using C To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Thu, 8 Mar 2007, Wai Wu wrote: You are right. However, he still has to have some kind of user interface or an external application to talk to that code or have the code directly poll numbers from a database. I say just use a scheduler to create the .call files since as soon as the files get to the outbound directory, Asterisk will just eat them. The documented interfaces (call files and AMI) are less likely to induce instability and are more likely to survive future releases as well. Both the callfiles and AMI APIs are reputed to lose requests (or perhaps other failures) under heavy loads. Is this still true in 1.4.x? Is there any current work on those bugs? Is there any benchmark on how much traffic is safe, 100% reliable? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] AJAM Documentation
Hi, I would to know where I can find some specific documentation on AJAM. Thanks Best regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] AJAM Documentation
see doc/ajam.txt On 3/9/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I would to know where I can find some specific documentation on AJAM. Thanks Best regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- Clod Patry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] how res speak with app ?
can we help me ? anyone can explain to me what I need to app menage to speak with res? regards. -- Mhayk Whandson da Silva Lima MSN: [EMAIL PROTECTED] ICQ: 163967537 GoogleTalk: [EMAIL PROTECTED] Skype: mhaykwhandson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] disable client side hangup after dialing 911on SIP Dial
Hi I think it would be a good feature to be able to reproduce the following behavior that a standard phone line does with 911. Normally if someone calls 911 and hangs up after the call has been established then the line is not dropped because it is held by the 911 agent. If you pickup your phone you should still be connected to the 911 agent and be able to talk to him. The call is dropped only when the 911 agent hangs up on his side. Maybe asterisk could disable the hangup from the client after he has dialed 911 Or maybe asterisk can keep the channel up and call back the user to re-establish the call until the hangup comes from the other side I am talking about a SIP Dial and not a Zap Dial, it seems that this option is available on Zap Dial. Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] disable client side hangup after dialing 911on SIP Dial
Admin DeryTelecom wrote: I am talking about a SIP Dial and not a Zap Dial, it seems that this option is available on Zap Dial. Then this will be the responsibility of the SIP endpoint, not Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] disable client side hangup after dialing 911on SIP Dial
On Friday 09 March 2007 13:58, Kevin P. Fleming wrote: Admin DeryTelecom wrote: I am talking about a SIP Dial and not a Zap Dial, it seems that this option is available on Zap Dial. Then this will be the responsibility of the SIP endpoint, not Asterisk. It cannot. It is a violation of the SIP specification for a UAS to act in this way, and therefore, the UAC has no requirement to work that way, either. See RFC 3261, section 15.1.2: A UAS core receiving a BYE request for an existing dialog MUST follow the procedures of Section 12.2.2 to process the request. Once done, the UAS SHOULD terminate the session (and therefore stop sending and listening for media). The only case where it can elect not to are multicast sessions, where participation is possible even if the other participant in the dialog has terminated its involvement in the session. Whether or not it ends its participation on the session, the UAS core MUST generate a 2xx response to the BYE, and MUST pass that to the server transaction for transmission. In other words, the server is not permitted to deny the BYE if it is properly formatted and within an existing dialog. -- Tilghman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Bug Squashing Meeting
This is a reminder that Digium will have an open conference call on Wednesday (3/14) at 1 pm Central Time. The purpose of this meeting will be a status update on the 15 mantis issues that were selected last week. 15 issues will be discussed, so the closed issues will be replaced with the next oldest issues in the 'new' state. Anybody interested in joining the conference call can participate via: IAX2/[EMAIL PROTECTED]/asterisk-dev Thanks! Dwayne Hubbard Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Re: redirecting a call after a Queue
On 09 Mar 2007 13:30:28 +0100, Benny Amorsen [EMAIL PROTECTED] wrote: DB == Dov Bigio [EMAIL PROTECTED] writes: DB So, the idea would be to have a parameter for the Queue app that DB would be the extension to transfer the call to in case the call is DB finished by the agent. You can catch the call in the h extension (with a Goto). We have been playing with that; unfortunately not with complete success. Since the call hasn't been hung up, can't you just catch it on the next priority? By the time it hits h, you'll already be hungup, and playing audio doesn't work well from that state ;) --Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
