[Freeswitch-users] Jack and portaudio support in freeswitch
Hello, are there any plans to make freeswitch compatible with jack and puredata, like russel has done for asterisk ( http://www.russellbryant.net/blog/2008/01/13/jack-interfaces-for-asterisk/) . Such a setup will enable us to manipulate or play with rtp in realtime like (http://www.lobstertech.com/code/voicechanger/) . Since both jack and puredata have a posix c++ interface, it shudnt be difficult. cheers!! ashutosh ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GUI
Hi, Darren has released the pre-alpha release of the Freeswitch GUI. Check out more at http://www.d-man.org/news/2008/09/03/83 . Though waiting for an announcement from him officially at this mailing list. Looking forward! Cheers!! ashutosh On Wed, Aug 13, 2008 at 12:50 AM, Anthony Minessale [EMAIL PROTECTED] wrote: it's coming soon. On Tue, Aug 12, 2008 at 9:40 AM, Nicolas Brenner [EMAIL PROTECTED]wrote: On Fri, Aug 1, 2008 at 9:51 AM, Anthony Minessale [EMAIL PROTECTED] wrote: ... A quad woodcrest 2.6ghz can do about 3000 simo media sessions with FS, the same box can just make it to 400 when they are all G729 transcoding calls. If they are bridged calls, that number goes in half, if we take media out of the picture that number quadruples. So I guess I could boast 400 CPS with 3000-6000 simo sessions, but what's the point, I'll let Ken do that.. ;) G729 transcoding? I thought there was no support for that... Anyway I can get it (other than writing it myself)? -- Nicolás Brenner ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Bug in spidermonkey-odbc-numRows
Hi,
i observed that mod_spidermonkey_odbc function numRows has been returning
me number of columns rather than rows. Looking deep revealed following:
file :
freeswitch-1.0.1/src/mod/languages/mod_spidermonkey_odbc/mod_spidermonkey_odbc.c
*
static JSBool odbc_num_rows(JSContext * cx, JSObject * obj, uintN argc,
jsval * argv, jsval * rval)
{
..
..
if (odbc_obj-stmt) {
SQLNumResultCols(odbc_obj-stmt, rows);}
.
.*
I suppose the fn call in bold above might be problematic. Has anyone else
countered it before ?I will try to resolve and put the patch if resolved.
Thanks,
ashutosh
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Re: [Freeswitch-users] Documentation suggestion
Hi, I almost visit the docu wiki of FS daily to see if some documentation has been changed/added or altered. So, i have to literally go through all the sections and sub-sections of the wiki to find material of interest which might have changed in last 24 hours. I wondered if the home page of the wiki has a seciton like NEWS which can contain chronological changes in the docu wiki made by users/maintainers. That will enable us to find changes at one go. I will do my bit of study on mediawiki if this is possible and lchk if this is possible. Thanks to the FS team for the telephony of the century :) Cheers! -Ashutosh On Wed, Aug 6, 2008 at 10:10 AM, Robert Smith [EMAIL PROTECTED] wrote: Michael Collins wrote: And thank you! We appreciate it when people make suggestions about documentation. Everyone wants the program to do something but precious few people offer feedback on getting the system documented. Please continue offering suggestions. I speak as someone new to telephony applications, being asked to investigate them for our business model. As I (we) are researching software switches and IVRs we naturally installed Asterisk but found a problem. While searching for answers we found a list of soft switches on voip-info which led here. However, the learning curve is substantial to someone outside of telephony. May I suggest that someone provides a simple to understand page listing call flow from a PSTN and out to a PSTN, for example? Simply working out what functionality exists and how it is intended to be used would make an excellent starting point. Simple things like explaining the difference between users and extensions. Agreed much of this is not your responsibility. However for take-up to be effective by a wide audience such guidance would be a serious boost. R. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Documentation suggestion
Ohhh.. i see that. Thanks for pointing it. :) Cheers! -ashu On Wed, Aug 6, 2008 at 6:24 PM, Brian West [EMAIL PROTECTED] wrote: On the left side click recent changes. That should help greatly. /b Sent from my iPhone On Aug 6, 2008, at 1:13 PM, Ashutosh [EMAIL PROTECTED] wrote: Hi, I almost visit the docu wiki of FS daily to see if some documentation has been changed/added or altered. So, i have to literally go through all the sections and sub-sections of the wiki to find material of interest which might have changed in last 24 hours. I wondered if the home page of the wiki has a seciton like NEWS which can contain chronological changes in the docu wiki made by users/maintainers. That will enable us to find changes at one go. I will do my bit of study on mediawiki if this is possible and lchk if this is possible. Thanks to the FS team for the telephony of the century :) Cheers! -Ashutosh On Wed, Aug 6, 2008 at 10:10 AM, Robert Smith [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Michael Collins wrote: And thank you! We appreciate it when people make suggestions about documentation. Everyone wants the program to do something but precious few people offer feedback on getting the system documented. Please continue offering suggestions. I speak as someone new to telephony applications, being asked to investigate them for our business model. As I (we) are researching software switches and IVRs we naturally installed Asterisk but found a problem. While searching for answers we found a list of soft switches on voip-info which led here. However, the learning curve is substantial to someone outside of telephony. May I suggest that someone provides a simple to understand page listing call flow from a PSTN and out to a PSTN, for example? Simply working out what functionality exists and how it is intended to be used would make an excellent starting point. Simple things like explaining the difference between users and extensions. Agreed much of this is not your responsibility. However for take-up to be effective by a wide audience such guidance would be a serious boost. R. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.orghttp://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Comparison matirx
Hi Simon, Though i hold the view that freeswitch will be winning hands up in most of categories, except security flaws and fix-around time, since we havent had any security flaws in FS till now...YET its my view that FS shouldnt be brought up for benchmarking against other solutions just yet. SipX and Asterisk have been around since long, and widely adopted, so more n more people can comment about them, but FS ain't seen the light of the day yet...so its too premature to put it in the fighting ring right now. Of course, thats my personal view, and someone can differ. An user base of at least .5 million user base should be enough (of which at least 10% are sort of gurus) would be enough when this matrix can be put up. Cheers!! Ashutosh On Sun, Aug 3, 2008 at 1:33 AM, UV [EMAIL PROTECTED] wrote: Excellent and important initiative, IMHO. I believe FreeSWITCH have sufficient objective advantages compared to not only open-source solutions, but commercial ones as well. I think that it can stand out even in a non-biased matrix. Take a look at these comparisons as a reference: http://www.voip-news.com/whitepaper/voip-ip-pbx-comparison/ http://www.networkworld.com/buyersguides/guide.php?cat=877966pcw_bg=mt=com pare http://coreg.tmginteractive.com/display/FullPage/Campaign943/OpenWhitePaper . aspx But it all depends on the purpose of the matrix in the first place. Just a thought. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Skopis (Lists) Sent: Sunday, August 03, 2008 11:06 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Comparison matirx Grey Man wrote: [snip] One suggestion I'd have for another row is Security Fix Rate. For example while the Asterisk community's approach to handling security releases is commendable the rate at which they happen is a real pain when you have to potentially upgrade a production system for each one. Although the pain comes from having to worry about whether the version of Asterisk that you need to upgrade to will be one of the stable or dud versions! I would certainly agree the security is important. Responsiveness to security flaws is one thing. I think another point of valuation would be average bugs per year or month, weighted accordingly (pre-auth remote command execution should have a greater weight than an xss in the built-in web server). Though, that might turn into a whole other book. ;] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.5.10/1586 - Release Date: 01/08/2008 18:59 No virus found in this outgoing message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.5.10/1587 - Release Date: 02/08/2008 17:30 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to reconnect to FS ?
I wondered if attaching FS to a a screen and in bg mode would be good workaround, so that one can reattach to that screen from any other shell later ? Regards, ashutosh On Fri, Aug 1, 2008 at 1:32 AM, Diego Viola [EMAIL PROTECTED] wrote: I know there is fs.pl, and it's indeed useful, but an -r option in the binary itself would be nicer ;-) Just as a suggestion, don't take it bad =D. FS rocks! Diego On Thu, Jul 31, 2008 at 3:57 PM, Henk Oegema [EMAIL PROTECTED] wrote: On Thursday 31 July 2008 20:54:53 unknown wrote: freeswitch# perl fs.pl FreeSWITCH That's it. :-) Henk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Scripting Language
Hi brothers, Now that Freeswitch is there, the only problem remains is of choosing the right language to use with it, as i hold the view, that you can get the best from FS only by using the best and efficient call control language . I am not sure about the following. Please clarify. -For all call control languages (CCL) like js, lua, perl, python, do many calls coming in spawn one instance of the script (a.k.a static class), or each call spawns its own instance of script? -if the answer to above is yes, then is there CCL among lua,perl,python, which has the ability to spawn a single instance, and open and create child threads at will, for all incoming calls ? This is of concern since its essentially bad to have an excellent softswitch, and an inefficent and memory-hogging script working with it. -If there is indeed a CCL to do the above, am i right in saying that the combination of this particular language with FS will yield the highest number of CPS on any given box, (with and without transcoding), compared to other CCLs. Though personally, my choice is of using mod_perl for the above reason because of its daemonized nature, but if lua offers the same, i m willing to spend few nights learning lua as well. Please shed light someone. Thanks for all you do! -Ashutosh ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] using freeswitch for high volume call traffic
Hi brothers, Now that Freeswitch is there, the only problem remains is of choosing the right language to use with it, as i hold the view, that you can get the best from FS only by using the best and efficient call control language . I am not sure about the following. Please clarify. -For all call control languages (CCL) like js, lua, perl, python, do many calls coming in spawn one instance of the script (a.k.a static class), or each call spawns its own instance of script? -if the answer to above is yes, then is there CCL among lua,perl,python, which has the ability to spawn a single instance, and open and create child threads at will, for all incoming calls ? This is of concern since its essentially bad to have an excellent softswitch, and an inefficent and memory-hogging script working with it. -If there is indeed a CCL to do the above, am i right in saying that the combination of this particular language with FS will yield the highest number of CPS on any given box, (with and without transcoding), compared to other CCLs. Though personally, my choice is of using mod_perl for the above reason because of its daemonized nature, but if lua offers the same, i m willing to spend few nights learning lua as well. Please shed light someone. Thanks for all you do! -Ashutosh On Tue, Jul 29, 2008 at 5:48 AM, Ruchir Brahmbhatt [EMAIL PROTECTED] wrote: Hi Everyone, I'm evaluating various solutions for a requirement of softswitch. Main requirements are support for many simultaneous calls, g729 g723 codecs, CDR, billing, etc. I was wondering if freeswitch is the right solution. I heard that it can handle many simultaneous calls and calls per second is also good. But i have doubts about CDR and billing. Can it do prepaid? Only java script is supported as scripting language? Can it write CDR in mysql table? Can it get configuration from mysql db(like asterisk realtime)? Does it stay in media path when both ends are using g729 or g723 codecs and freeswitch is doing it in passthrough mode? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Equivalent of asterisk ID
Hi, I have been looking for an attribute in the FS CDR which will be unique for a call, like we had uniqueid field in asterisk cdr. Someone have any idea ? Thanks, ashutosh ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Freeswitch-dev] YAML support as an alternative of XML for configuration
Anthony rocks, FS team rocks, will keep rocking!! 3 hour release show a good practice of agile programming at its best. Essentially, asterisk was not bad; actually, the fact is that FS is so so much outstanding, cleaner and working ,that asterisk pales out. Frankly, after seeing this post thread, i lost respect to digium; they turned into a corp from an opensource product, and introduced unneeded hierarchies et al, which decelerated asterisk development, and increased release time, that too unstable , and with bugs. No one there owns up bugs in mantis, let alone deciding on resolution. Salutes to the never ending spirit of man, thats called Anthony!! Countdown to asterisk extinction ? Cheers to all ! Ashutosh On Tue, Jul 1, 2008 at 5:02 AM, Ghulam Mustafa [EMAIL PROTECTED] wrote: Diego, This is sad, you claim that FreeSWITCH is more advanced in terms of features and innovation than Asterisk, but all I see with your attitude is that it's more limited and closed-minded. how come they are close-minded and limited, when open source(e.g freeswitch) gives you freedom to modify code for your own needs/ease. why don't write a mod_yaml? ;) On Mon, 2008-06-30 at 20:24 -0400, Diego Viola wrote: Read more about it here: http://en.wikipedia.org/wiki/YAML Diego On Mon, Jun 30, 2008 at 7:14 PM, Diego Viola [EMAIL PROTECTED] wrote: This rocks! Anthony: Thank you so much for this... and sorry for the troubles I caused, I do not think you or anyone else in your team is a closed minded person... FreeSWITCH is the best fruit of your great talent and open mind ;-) Diego On Mon, Jun 30, 2008 at 4:43 PM, EdPimentl [EMAIL PROTECTED] wrote: Net Net, anyway you want to say... this is an amazing team. In XML: open source=AmazingTeam on=FreeSwitch projectownerAnthony/projectowner developmentMike/development qaBrian/qa /open In YAML: AmazingTeam: project : FreeSwitch projectowner: Anthony projectlead: Mike qa: Brian ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ghulam Mustafa [EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Equivalent of asterisk ID
Hi Michael, Will two legs of the same call show up the same uuid in the cdr ? Thanks, ashutosh On Wed, Jul 23, 2008 at 7:37 PM, Michael Collins [EMAIL PROTECTED] wrote: Is there any reason that the uuid field doesn't work for you? -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Ashutosh *Sent:* Wednesday, July 23, 2008 12:18 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Equivalent of asterisk ID Hi, I have been looking for an attribute in the FS CDR which will be unique for a call, like we had uniqueid field in asterisk cdr. Someone have any idea ? Thanks, ashutosh ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Freeswitch-dev] YAML support as an alternative of XML for configuration
Yeah, That was implied in the very first line FS Team rocks! Cheers again all!! -ash On Wed, Jul 23, 2008 at 7:56 PM, Gonzalo Servat [EMAIL PROTECTED] wrote: On Wed, Jul 23, 2008 at 4:34 PM, Ashutosh [EMAIL PROTECTED] wrote: [..snip..] Salutes to the never ending spirit of man, thats called Anthony!! .. and the entire FS team! - Gonzalo ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Equivalent of asterisk ID
ok, is there any way or a common variable through which i can relate calls of the same id. What does the field sip_call_id field do ? Best regards, ashutosh On Wed, Jul 23, 2008 at 7:57 PM, Brian West [EMAIL PROTECTED] wrote: no each leg has its own uuid. /b On Jul 23, 2008, at 2:41 PM, Ashutosh wrote: Hi Michael, Will two legs of the same call show up the same uuid in the cdr ? Brian West sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Timeout parameter in originate
Hi, We see originate function as follows: result = new_session.originate(session, dest[[, dialplan], context], cid_name], cid_num], network_addr], ani], aniii], rdnis], username], to]); Where to= timeout When does the timeout counter start, after call to the originate Api, or after the remote end has picked up the call ? Thanks, ashutosh ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Playing an audio file to end destination before bridging the call
Hi,
So, you want call from A to B, and make B listen to a file when he picks up,
right ?
You can originate the call to LegB , and make it drop to a context which
does the following
- first play the file
- Then bridge to LegA
Regards,
ashutosh
On Mon, Jul 14, 2008 at 8:00 AM, Adnan Barakat [EMAIL PROTECTED]
wrote:
Joseph Bajin wrote:
I assume you are meaning as you are connecting, you may want to play a
custom ringback or fake the ring.
Here's the page to do it:
http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
Or if you are using the originate command:
originate {ringback=/path/to/music.wav}sofia/gateway/name/number
bridge(sofia/gateway/name/othernumber)
Thanks Joe, but this will play the file to the caller, I'm trying to play a
file
to the receiver (ie. when the receiver picks up the phone the voice file
will be
played to them [the receiver], then the call will get bridged - and while
the
file is being played [to the receiver] the caller will continue to hear the
normal ringtone until the call is bridged)
Adnan
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
[Freeswitch-users] Background music in conversation
Hi, The scnario required is as follows: A calls B using FS, and a music should be played in the background while they are talking. I have searched whole wiki and fs docs and mod_dptools but unable to find a pointer so far. Even in asterisk, this has no plain solutions, other than to bring both the parties in a meetme room and starting moh for a third party on local/ channel. Anyone help me out or any links ? Best Regards, ashutosh ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
