Hi All, I once meet the Skypiax load error issue and some guys infomed me that there is no configuration file for Skypiax. When I follow these intructions -> http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to_... I still have some difficulties unstanding this: ." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to Debug\conf\autoload_configs." I did not find such directories in my freeswitch folder. Did not understand what "src" means, I checked the freeswitch folder and did not find such a folder named"src". There is a folder named"mod" under freeswitch while look into "mod" folder there are only some DLL files and can not find endpoints and etc. 2.You'll probably build the "Debug" version I just build this from the precompiled binaries and then launched FS. I'm not sure what I launched is in debug mode or not. If anyone can offer some help that really be appriciated. Thanks Daniel Zeng From: freeswitch-users-request@lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 40 To: freeswitch-users@lists.freeswitch.org Date: Thu, 5 Nov 2009 06:57:44 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request@lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --附转发的邮件-- From: anthony.minessale@gmail.com To: freeswitch-users@lists.freeswitch.org Date: Thu, 5 Nov 2009 08:13:46 -0600 Subject: Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media I did not ask you to send me a ladder diagram. I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me) 1) start FreeSWITCH 2) run the cli command: console loglevel debug 3) run the cli command: sofia profile internal siptrace on 4) reproduce your issue and put the trace on freeswitch pastebin http://pastebin.freeswitch.org (login and pass are stated in the auth dialog) Also please answer brian's question. What phones and/or sip devices are involved in this call. On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana <hjqlopez@hotmail.com> wrote: Thanks for your time, -The scenario is still the same: Always bypass media. Environment 100% NAT free :-) Call established from A to B through FS. Then... Blind transfer from B to C (Refer-to: C) RTP should go directly between A and C. -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the lack of reINVITE to A, after C answers). Please check SIP diagram here: http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html -What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-) http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html -You could check the sofia debug for r15332 here: http://pastebin.com/m6f2b3836 Best regards, Humberto
I don't know what you are talking about anymore.
The scenario I had tested is when a call is bridged in bypass_media=true
bridge
and you blind transfer that call back to the dialplan
as soon as it hits the routing state it will resume media.
it has been confirmed to not work and confirmed to have been fixed several
time and if you are still having a problem you must have something blocking
some of your packets or something .
You have to understand that sip is a protocol and your description is
completely non-standard.
Perhaps you should get a console trace and attach it to a jira. The trace
probably makes more sense to me.
sofia profile internal siptrace on
console loglevel debug
reproduce and attach the whole capture.
On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote:
Hi,
I tried r15332 and set in the sofia profile:
a) bypass_media_after_bridge=true only
b) bypass_media_after_bridge=true, param name="media-option"
value="resume-media-on-hold"/>
In both cases FS is hanging up the initial call (A to FS) after accepting
the REFER to C:
A <- reINVITE with FS' SDP <- FS
A -> 200 -> FS
A <- ACK <- FS
A <- BYE <- FS
The call to C is not even tried.
I found this line is the logs that could give some idea:
2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup
sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE]
after sending the ACK for the reINVITE
Regards,
Humberto
please try r15326
I think i have it working.
I recommend for optimal results you set bypass_media_after_bridge=true
either as a global or in your DP in place of bypass_media=true
On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana
hotmail.com>wrote:
Hi Mike,
I re-tried with trunk rev 15319 but I got almost the same behavior:
There
is now a reINVITE (with FS' SDP) going to A when the REFER is accepted.
But
still there is no reINVITE for A (with C's SDP) after the call from FS
to C
is established.
Anyway, we decided for now to do a different implementation but if you
want
to explore more in this issue count me in ;-)
Thank you very much!
Humberto
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_________________________________________________________________ Windows Live: Make it easier for your friends to see what you’re up to on Facebook. http://go.microsoft.com/?linkid=9691816 _______________________________________________ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 --附转发的邮件-- From: rob4manhere@gmail.com To: freeswitch-users@lists.freeswitch.org Date: Thu, 5 Nov 2009 08:52:05 -0600 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hi UK, From what I've done and read, the caller-controls (in conference.conf.xml) can be modified to almost anything you can think of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a caller control just for those with the moderator flag. So unless you want everyone able to mute/kick everyone then you can't do it. The wiki seems to indicate this as well: "Be aware that the caller-controls are applied across the entire conference. You cannot enter one member of the conference using caller- controls ABC and then enter a second member using caller-controls XYZ." http://wiki.freeswitch.org/wiki/Mod_conference I think this might be a limitation of mod_conference. Perhaps one of the pros can chime in if I'm off-base or there's some nifty way to accomplish this. Cheers, Rob On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote:
Any ideas on the below...has anyone implemented the below:
Once I have the Moderator and Participants logged on, how do I invoke the moderator previlidges, LIk esay muting everyone/someone or kicking someone out of the Conf and the like?
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Ujjval Karihaloo Sent: Monday, November 02, 2009 12:52 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Rob:
Once I have the Moderator and Participants logged on, how do I invoke the moderator previlidges, LIk esay muting everyone/someone or kicking someone out of the Conf and the like?
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Rob Forman Sent: Friday, October 30, 2009 9:34 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Hm, strange. I haven't seen that before. Can you pastebin your logs at debug level?
On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
It's strange... a tcpdump tells me that there is no DTMF from my provider when using IVR, but when I call into a TN that goes directly into the Conference App, I see DTMF from the provider.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Rob Forman Sent: Friday, October 30, 2009 7:23 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I've never had any problem with that. Is your logging at debug level so you can see the RECV DTFM in the log/fs_cli? Are you calling from a SIP phone on the pbx, or via a PSTN provider? Maybe your provider isn't passing them through.
Make sure your logging is turned up then try something simpler, like calling the echo application, and see if DTFM comes through.
Rob
On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote:
Rob:
For some reason, I don't see the DTMF appear on the fs_CLI when using the below configuration....so it basically timesout.
UK
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Ujjval Karihaloo Sent: Monday, October 26, 2009 9:21 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
Thx a lot Rob, reading the wiki your way or using IVR seems correct.. =============== The wiki also says that the wait-mod might be "used in conjunction with an IVR where the moderators are authenticated with an extra pass- code", which is what I did. I guess that's why I didn't understand the point of the +pin. ======================
I will try it out.
Again thx a lot for your help. Will keep everyone posted.
________________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Rob Forman [rob4manhere@gmail.com] Sent: Friday, October 23, 2009 12:22 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator
I just re-tested with the pin in my dial plan:
<action application="conference" data="conference 123456@default +flags{}+1234" />
And it doesn't challenge me for the pin. I just drop right in. I figured this is how it was intended, since the wiki says the pin is set initially and only challenged in later attempts [by future callers]:
"The first time a conference name (confname) is used, it will be created on demand, and the pin will be set to what ever is specified at that time: the pin in the data string if specified, or if not, the "pin" setting in the conference profile, and if that is also unspecified, then there is no pin protection. Any later attempt to join the conference must specify the same pin number, if one existed when it was created. "
The wiki also says that the wait-mod might be "used in conjunction with an IVR where the moderators are authenticated with an extra pass- code", which is what I did. I guess that's why I didn't understand the point of the +pin.
I'm sure there's a scenario where its used and useful, the wiki just doesn't explain it.
Rob
On Oct 23, 2009, at 12:43 PM, Brian West wrote:
Well first off you're not defining a pine here...
confname@profilename+flags{mute|deaf|waste|moderator}+[conference pin number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
<entry action="menu-exec-app" digits="1" param="conference 123456@default+flags{}" />
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--附转发的邮件-- From: rob4manhere@gmail.com To: freeswitch-users@lists.freeswitch.org Date: Thu, 5 Nov 2009 08:57:08 -0600 Subject: [Freeswitch-users] Wideband / HD phones Hey all, Looking at buying some high def phones. Any recommendations (preferably based on experience) for hardware based on product quality, standards compliance, features integration with Freeswitch, etc? Thank you! Rob Forman _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1