Re: [Freeswitch-users] How to upgrade ?
It might be premature as well as half baked - I am just testing out the thought, however here is my thinking... There is a folder called extensions in the dialplan folder. The nice thing about this folder is that it is read in by a simple xml preprocessing command that says go get *.xml Currently, it is being called from the Default dialplan. Borrowing from the Apache HTTPD style configuration, everything to customize the install is done with additional files as opposed to editing anything that ships with the software. So This works, however, we should be able to specify the name of a root configuration folder - this way all config can sit way outside of the app tree - for example, to place the FS app in one folder and have all external config in a different folder not under the same path - thereby ensuring upgrade safety. My patch is only going to work with dialplans, but I propose making it a pattern to follow. In the dialplan, I am making a folder for each context that will be loaded externally and postfixing a .d on the end, so for example default.d and public.d Make sense? Michael Jerris wrote: I don't understand the proposal. Could you explain a bit more please. Mike On Sep 20, 2008, at 6:39 PM, Christian Jensen [EMAIL PROTECTED] wrote: I am working on a patch that would externalize the install configs, much the same way as the extensions folder (which i propose to be called default) The new folders will be the same name as the context they are called from, for example the one i am most interested in is public.d Any objections? -Original Message- From: Michael Jerris [EMAIL PROTECTED] Sent: September 20, 2008 3:24 PM To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Cc: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to upgrade ? Make current will not work from release tarballs but if you grab the new version it should install without wiping out your config. Mike On Sep 20, 2008, at 6:10 PM, Michael S Collins [EMAIL PROTECTED] wrote: Yes make current will update you without breaking your existing configs. Even if you do make samples it won't overwrite your configs bit it will replace any missing conf files with default ones. -MC Sent from my iPhone On Sep 20, 2008, at 2:46 PM, henkoegema [EMAIL PROTECTED] wrote: At the moment I'm using FS version 1.0.0. I want to upgrade to the latest version. Can I just type #make current(?):working: without loosing my own (or edited standard) conf files ? -- View this message in context: http://www.nabble.com/How-to-upgrade---tp19589317p19589317.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 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
Re: [Freeswitch-users] How to upgrade ?
This was what we talked about on IRC a few days ago right? /b On Sep 21, 2008, at 2:45 AM, Christian Jensen wrote: It might be premature as well as half baked - I am just testing out the thought, however here is my thinking... There is a folder called extensions in the dialplan folder. The nice thing about this folder is that it is read in by a simple xml preprocessing command that says go get *.xml Currently, it is being called from the Default dialplan. Borrowing from the Apache HTTPD style configuration, everything to customize the install is done with additional files as opposed to editing anything that ships with the software. So This works, however, we should be able to specify the name of a root configuration folder - this way all config can sit way outside of the app tree - for example, to place the FS app in one folder and have all external config in a different folder not under the same path - thereby ensuring upgrade safety. My patch is only going to work with dialplans, but I propose making it a pattern to follow. In the dialplan, I am making a folder for each context that will be loaded externally and postfixing a .d on the end, so for example default.d and public.d Make sense? ___ 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] Possible problem in adding channel variables to the bridge
The FREESWITCH version is the 1.0.1 release version. I have updated to the snapshot from the 20th of September. But I get problems with the ./configure. The bootstrap.sh works OK after I have updated autoconf to version 2.63 and automake to version 1.10.1. The error from ./configure after half a page of lines is: ./configure: line 4693: syntax error near unexpected token `build_libtool_libs,' ./configure: line 4693: `_LT_DECL(build_libtool_libs, enable_shared, 0, Operating system is CentOS, uname -r = 2.6.9.-67.0.4.plus.c4. Jon ___ 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] cant record any session
i just copied the dialplan given in wiki for record session but cant record or lets say first i cant initiate the call only, sofia says codec not found as soon as i hit dial on the dialer can some1 help in the xml as im trying to record calls made from a specific user whose account is created only and not in general for every1 -- View this message in context: http://www.nabble.com/cant-record-any-session-tp19594481p19594481.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] Possible problem in adding channel variables to the bridge
Try automake 261 /b On Sep 21, 2008, at 8:16 AM, Jon Bruel wrote: The FREESWITCH version is the 1.0.1 release version. I have updated to the snapshot from the 20th of September. But I get problems with the ./configure. The bootstrap.sh works OK after I have updated autoconf to version 2.63 and automake to version 1.10.1. The error from ./configure after half a page of lines is: ./configure: line 4693: syntax error near unexpected token `build_libtool_libs,' ./configure: line 4693: `_LT_DECL(build_libtool_libs, enable_shared, 0, Operating system is CentOS, uname –r = 2.6.9.-67.0.4.plus.c4. Jon ___ 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] Wrong port on response
The Aastra 55i should have STUN support, but not on the firmwares released with the phone initially. Check your firmware version here: http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-FE4A1E69/03/hs.xsl/21 669.htm For the 480i, though, I think you're SOL. - Darren -Original Message- From: Brian West [mailto:[EMAIL PROTECTED] Sent: Friday, September 19, 2008 4:35 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Wrong port on response Yep just checked in my 55i that I have and no STUN nor RPORT. Great for the LAN silly for the WAN. the force-rport is your only option and its a global per profile setting. /b On Sep 19, 2008, at 6:08 PM, David Aldworth wrote: User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/ v3.2.8.45. ___ 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] Wrong port on response
I have looked all over the web interface and can't find it. I have the latest firmware too... weird. Still no rport support..?!? /b On Sep 21, 2008, at 11:45 AM, Darren Schreiber wrote: The Aastra 55i should have STUN support, but not on the firmwares released with the phone initially. Check your firmware version here: http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-FE4A1E69/03/hs.xsl/21 669.htm For the 480i, though, I think you're SOL. - Darren ___ 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] Conference members variable
Hi Everybody I am building a conferencing service, and I was wondering if there is a system variable that I can use, that holds the number of members in a specific conference. Is there a variable I can read (in java, or anywhere) that tells me this ? If there is, is there a list of FreeSWITCH variables anywhere ? I can read the members (and the total) via CLI (conference conference list), but I don't know how to import this figure to my (java)script. Thank you very much for your time Torstein ___ 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] Wrong port on response
Hi Brian, I have Aastra 55i, it does have STUN and Rport support, and I like its rport support. On web interface, under network settings-Advanced Network Settings-Rport (RFC 3581)Enabled for Rport or STUN Server STUN Port for STUN support, however I like rport better, which I can have call forwarding working seamlessly for freeswitch in public ip address, even my Aastra is behind my doule-NATed home network. Best regards, Chris On Sun, Sep 21, 2008 at 12:51 PM, Brian West [EMAIL PROTECTED] wrote: I have looked all over the web interface and can't find it. I have the latest firmware too... weird. Still no rport support..?!? /b On Sep 21, 2008, at 11:45 AM, Darren Schreiber wrote: The Aastra 55i should have STUN support, but not on the firmwares released with the phone initially. Check your firmware version here: http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-FE4A1E69/03/hs.xsl/21 669.htm For the 480i, though, I think you're SOL. - Darren ___ 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] Exchange 2007 UM - DTMF problem
you can see in the trace that the exchange is using alaw. So the complaint about trans coding is because it's going from PCMU on leg a to PCMA when talking to exchange. please revert any changes, upgrade to current SVN trunk and try configuring exchange to use PCMA or confiigure FS to only use PCMA by editing vars.xml and making the codec_string be PMCA instead of PCMU On Sat, Sep 20, 2008 at 10:08 PM, Matt Darnell [EMAIL PROTECTED]wrote: In the meantime, Matt, try applying this non-standard change (it worked for me). Change src/switch_rtp.c (lines 1197-1198) from: rtp_session-dtmf_data.out_digit_sofar = samples; rtp_session-dtmf_data.out_digit_sub_sofar = samples; to: rtp_session-dtmf_data.out_digit_sofar = 0; rtp_session-dtmf_data.out_digit_sub_sofar = 0; Let me know if that fixed your DTMF problem with your UM. I made the change, did configure, make clean, make, make install. I see this in the log now: 2008-09-20 16:54:32 [DEBUG] switch_rtp.c:1201 do_2833() Send start packet for [3] ts=12560 dur=0/0/2000 seq=15883 Before the duration would start at 160/160/2000. Is that what I should see? It still doesn't solve the issue. It doesn't act like I would expect. The server either 'hears' digits on a call or it doesn't; I would expect it to miss digits every now and then, not either completely pass or completely fail. What type of DTMF failure were you experiencing? I wonder if it is not a Freeswitch issue but an Exchange issue, I will get a SIP/TCP softphone and try to connect directly to the Exchange server. I think SJPhone does SIP/TCP. -Matt ___ 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] Creating and destroying local_stream dynamically
Hi, I need to create and destroy local_streams dynamically, that is, I need to be changing the MOH of several fifo's in real-time, since local_stream creates a thread that is broadcasting audio I would like to be able to create and destroy them as I need them. How can this be achieved? Cesar Cepeda. ___ 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] Creating and destroying local_stream dynamically
This is not currently possible. It's something that could be added but would require a rework of mod_local_stream Mike On Sep 21, 2008, at 3:15 PM, Cesar Cepeda wrote: Hi, I need to create and destroy local_streams dynamically, that is, I need to be changing the MOH of several fifo’s in real-time, since local_stream creates a thread that is broadcasting audio I would like to be able to create and destroy them as I need them. How can this be achieved? Cesar Cepeda. ___ 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] cant record any session
ok i marked bypass media as well as proxy media to default which was like commented, means marked as comment. With proxy media to enabled i used to get the error of cant find codec but after making it all to default config, im now getting this 2008-09-22 00:07:46 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening Fi le [C:Program Files (x86)FreeSWITCH/recordings/2008-09-22-00-07-46_15194344406_b ipin.wav] [System error : The system cannot find the path specified. ] 2008-09-22 00:07:46 [ERR] switch_ivr_async.c:794 switch_ivr_record_session() Err or opening C:Program Files (x86)FreeSWITCH/recordings/2008-09-22-00-07-46_151943 44406_bipin.wav Brian West-3 wrote: Are you using bypass media? What exactly is the error? /b On Sep 21, 2008, at 8:29 AM, xbipin wrote: i just copied the dialplan given in wiki for record session but cant record or lets say first i cant initiate the call only, sofia says codec not found as soon as i hit dial on the dialer can some1 help in the xml as im trying to record calls made from a specific user whose account is created only and not in general for every1 -- View this message in context: http://www.nabble.com/cant-record-any-session-tp19594481p19594481.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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 -- View this message in context: http://www.nabble.com/cant-record-any-session-tp19594481p19597918.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] cant record any session
On Sep 21, 2008, at 4:11 PM, xbipin wrote: ok i marked bypass media as well as proxy media to default which was like commented, means marked as comment. With proxy media to enabled i used to get the error of cant find codec but after making it all to default config, im now getting this Correct, you can't do anything that interacts with the media if you are doing bypass or proxy modes. 2008-09-22 00:07:46 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening Fi le [C:Program Files (x86)FreeSWITCH/recordings/2008-09-22-00-07-46_15194344406_b ipin.wav] [System error : The system cannot find the path specified. ] Does it exist? 2008-09-22 00:07:46 [ERR] switch_ivr_async.c:794 switch_ivr_record_session() Err or opening C:Program Files (x86)FreeSWITCH/recordings/2008-09-22-00-07-46_151943 44406_bipin.wav Mike ___ 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] Creating and destroying local_stream dynamically
Thanks for an answer un Sunday ;) Cesar Cepeda. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Michael Jerris Enviado el: Sunday, September 21, 2008 2:59 PM Para: freeswitch-users@lists.freeswitch.org Asunto: Re: [Freeswitch-users] Creating and destroying local_stream dynamically This is not currently possible. It's something that could be added but would require a rework of mod_local_stream Mike On Sep 21, 2008, at 3:15 PM, Cesar Cepeda wrote: Hi, I need to create and destroy local_streams dynamically, that is, I need to be changing the MOH of several fifo's in real-time, since local_stream creates a thread that is broadcasting audio I would like to be able to create and destroy them as I need them. How can this be achieved? Cesar Cepeda. ___ 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] cant record any session
file doesnt exist but doesnt it need to create it by itself as u never know whose gonna call when and where. basically im looking for it to create the file and record in it so i can get a different file for each user, whats the point creating the file and then letting it record to it Michael Jerris wrote: On Sep 21, 2008, at 4:11 PM, xbipin wrote: ok i marked bypass media as well as proxy media to default which was like commented, means marked as comment. With proxy media to enabled i used to get the error of cant find codec but after making it all to default config, im now getting this Correct, you can't do anything that interacts with the media if you are doing bypass or proxy modes. 2008-09-22 00:07:46 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening Fi le [C:Program Files (x86)FreeSWITCH/recordings/2008-09-22-00-07-46_15194344406_b ipin.wav] [System error : The system cannot find the path specified. ] Does it exist? 2008-09-22 00:07:46 [ERR] switch_ivr_async.c:794 switch_ivr_record_session() Err or opening C:Program Files (x86)FreeSWITCH/recordings/2008-09-22-00-07-46_151943 44406_bipin.wav Mike ___ 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 -- View this message in context: http://www.nabble.com/cant-record-any-session-tp19594481p19598411.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] cant record any session
On Sep 21, 2008, at 5:00 PM, xbipin wrote: file doesnt exist but doesnt it need to create it by itself as u never know whose gonna call when and where. basically im looking for it to create the file and record in it so i can get a different file for each user, whats the point creating the file and then letting it record to it It said Path (not file) does not exist. Does the directory its trying to create the file in exist? ___ 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] cant record any session
yes it does Michael Jerris wrote: On Sep 21, 2008, at 5:00 PM, xbipin wrote: file doesnt exist but doesnt it need to create it by itself as u never know whose gonna call when and where. basically im looking for it to create the file and record in it so i can get a different file for each user, whats the point creating the file and then letting it record to it It said Path (not file) does not exist. Does the directory its trying to create the file in exist? ___ 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 -- View this message in context: http://www.nabble.com/cant-record-any-session-tp19594481p19598677.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] Wrong port on response
That would def. solve the problem :P /b On Sep 21, 2008, at 2:03 PM, unknown wrote: Hi Brian, I have Aastra 55i, it does have STUN and Rport support, and I like its rport support. On web interface, under network settings-Advanced Network Settings-Rport (RFC 3581)Enabled for Rport or STUN Server STUN Port for STUN support, however I like rport better, which I can have call forwarding working seamlessly for freeswitch in public ip address, even my Aastra is behind my doule-NATed home network. Best regards, Chris ___ 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] Creating and destroying local_stream dynamically
just keep the same directory name and change the contents of the directory as much as you want. On Sun, Sep 21, 2008 at 3:34 PM, Cesar Cepeda [EMAIL PROTECTED] wrote: Thanks for an answer un Sunday ;) Cesar Cepeda. *De:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *En nombre de *Michael Jerris *Enviado el:* Sunday, September 21, 2008 2:59 PM *Para:* freeswitch-users@lists.freeswitch.org *Asunto:* Re: [Freeswitch-users] Creating and destroying local_stream dynamically This is not currently possible. It's something that could be added but would require a rework of mod_local_stream Mike On Sep 21, 2008, at 3:15 PM, Cesar Cepeda wrote: Hi, I need to create and destroy local_streams dynamically, that is, I need to be changing the MOH of several fifo's in real-time, since local_stream creates a thread that is broadcasting audio I would like to be able to create and destroy them as I need them. How can this be achieved? Cesar Cepeda. ___ 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 -- 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] Transfer calls to internal user VS. dialing out
Hi,
I have a VERY simple FS setup.
1) Calls come in and go out via Vitelity. (I just route to their IP
address, don't even need a gateway.)
2) I have users registered remotely with anything from a single SIP
phone to an asterisk box.
3) Basically, I am a pass through of calls to and from Vitelity.
I have my user registering their phones on my default context.
Calls that come back in on a DID route through the public context.
Outgoing calls work perfectly.
My problem is with incoming calls. My users register with their DID
as their username.
So, when a call comes in on 323-555-1212, I want to route it to the
device registered with the username of 3235551212.
With my current setup, it is getting routed over to the default
context, and then FS is attempting to DIAL THE DID BACK OUT. which
creates a nasty loop.
HOW CAN I TELL FS TO ROUTE A CALL TO A REGISTERED USER AS OPPOSED TO
DIALING OUT???
My guess is that's its an adjustment to the transfer application in
my public context???
Thanks,
-Noah
-
I can call out just fine using the following in my dialplan:
context name=default
extension name=Vitelity
condition field=destination_number expression=^(1{0,1}\d{10})
$
action application=set
data=effective_caller_id_number=1222333/
!--action application=bridge
data=sofia/gateway/vitelity/$1/--
action application=bridge data=sofia/external/[EMAIL
PROTECTED]
/
/condition
/extension
/context
context name=public
extension name=inbound
condition field=destination_number
expression=^(3235551212)$
action application=transfer data=$1 XML default/
/condition
/extension
/context
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[Freeswitch-users] Gbridge
http://www.gbridge.com/ + FS = Awesome -E ___ 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] Transfer calls to internal user VS. dialing out
I would transfer to 1000 XML default for example. This is why you
split inbound and outbound into two context's like you have it... but
since you're sending the call into default as 10 digits I suspect it
loops right back out.
Also can you please not hijack threads.
Click new message, input the address yourself and then send it. If
you click reply, change the subject and body you're hijacking threads
DO NOT DO THAT please.
/b
On Sep 21, 2008, at 5:05 PM, Noah Silverman wrote:
HOW CAN I TELL FS TO ROUTE A CALL TO A REGISTERED USER AS OPPOSED TO
DIALING OUT???
My guess is that's its an adjustment to the transfer application in
my public context???
Thanks,
-Noah
-
I can call out just fine using the following in my dialplan:
context name=default
extension name=Vitelity
condition field=destination_number expression=^(1{0,1}\d{10})
$
action application=set
data=effective_caller_id_number=1222333/
!--action application=bridge data=sofia/gateway/vitelity/$1/
--
action application=bridge data=sofia/external/[EMAIL
PROTECTED]
/
/condition
/extension
/context
context name=public
extension name=inbound
condition field=destination_number
expression=^(3235551212)$
action application=transfer data=$1 XML default/
/condition
/extension
/context
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Re: [Freeswitch-users] How to upgrade ?
Yeppers. On Sep 21, 2008, at 12:52 AM, Brian West [EMAIL PROTECTED] wrote: This was what we talked about on IRC a few days ago right? /b On Sep 21, 2008, at 2:45 AM, Christian Jensen wrote: It might be premature as well as half baked - I am just testing out the thought, however here is my thinking... There is a folder called extensions in the dialplan folder. The nice thing about this folder is that it is read in by a simple xml preprocessing command that says go get *.xml Currently, it is being called from the Default dialplan. Borrowing from the Apache HTTPD style configuration, everything to customize the install is done with additional files as opposed to editing anything that ships with the software. So This works, however, we should be able to specify the name of a root configuration folder - this way all config can sit way outside of the app tree - for example, to place the FS app in one folder and have all external config in a different folder not under the same path - thereby ensuring upgrade safety. My patch is only going to work with dialplans, but I propose making it a pattern to follow. In the dialplan, I am making a folder for each context that will be loaded externally and postfixing a .d on the end, so for example default.d and public.d Make sense? ___ 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] Java script test
I have put the calltest.js in /usr/local/freeswitch/scripts and changed sofia to openzap/default/[EMAIL PROTECTED] in the coding and i got the error: Error: 2008-09-22 10:13:26 [ERR] switch_core_session.c:249 switch_core_session_outgoing_channel() Could not locate channel type openzap 2008-09-22 10:13:26 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED] 2008-09-22 10:13:26 [WARNING] mod_spidermonkey.c:2933 session_originate() Cannot Create Outgoing Channel! [openzap/default/[EMAIL PROTECTED] 2008-09-22 10:13:26 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! so i have changed that to sofia again. As Anthony Minessale said, i have changed @ symbol to % like sofia/default/1001%192.168.1.2 in the coding.But i am getting the following error: Error: 2008-09-22 10:18:08 [WARNING] mod_sofia.c:1949 sofia_outgoing_channel() Cannot locate registered user [EMAIL PROTECTED] 2008-09-22 10:18:08 [NOTICE] mod_sofia.c:2046 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2008-09-22 10:18:08 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [NO_ROUTE_DESTINATION] 2008-09-22 10:18:08 [WARNING] mod_spidermonkey.c:2933 session_originate() Cannot Create Outgoing Channel! [sofia/default/1001%192.168.1.2] 2008-09-22 10:18:08 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! Thanks. ___ 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] Java script test
Hi Preetha, you have use like this openzap/1/a/[EMAIL PROTECTED] Thanks Kris Anand On Mon, Sep 22, 2008 at 10:19 AM, preetha Ayyappan [EMAIL PROTECTED] wrote: I have put the calltest.js in /usr/local/freeswitch/scripts and changed sofia to openzap/default/[EMAIL PROTECTED] in the coding and i got the error: Error: 2008-09-22 10:13:26 [ERR] switch_core_session.c:249 switch_core_session_outgoing_channel() Could not locate channel type openzap 2008-09-22 10:13:26 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED] 2008-09-22 10:13:26 [WARNING] mod_spidermonkey.c:2933 session_originate() Cannot Create Outgoing Channel! [openzap/default/[EMAIL PROTECTED] 2008-09-22 10:13:26 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! so i have changed that to sofia again. As Anthony Minessale said, i have changed @ symbol to % like sofia/default/1001%192.168.1.2 in the coding.But i am getting the following error: Error: 2008-09-22 10:18:08 [WARNING] mod_sofia.c:1949 sofia_outgoing_channel() Cannot locate registered user [EMAIL PROTECTED] 2008-09-22 10:18:08 [NOTICE] mod_sofia.c:2046 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2008-09-22 10:18:08 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [NO_ROUTE_DESTINATION] 2008-09-22 10:18:08 [WARNING] mod_spidermonkey.c:2933 session_originate() Cannot Create Outgoing Channel! [sofia/default/1001%192.168.1.2] 2008-09-22 10:18:08 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! Thanks. ___ 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] Java script test
On Sep 22, 2008, at 12:49 AM, preetha Ayyappan wrote: I have put the calltest.js in /usr/local/freeswitch/scripts and changed sofia to openzap/default/[EMAIL PROTECTED] in the coding and i got the error: Error: 2008-09-22 10:13:26 [ERR] switch_core_session.c:249 switch_core_session_outgoing_channel() Could not locate channel type openzap 2008-09-22 10:13:26 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [openzap] cause: [CHAN_NOT_IMPLEMENTED] 2008-09-22 10:13:26 [WARNING] mod_spidermonkey.c:2933 session_originate() Cannot Create Outgoing Channel! [openzap/default/[EMAIL PROTECTED] ] 2008-09-22 10:13:26 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! mod_openzap isn't loaded. so i have changed that to sofia again. As Anthony Minessale said, i have changed @ symbol to % like sofia/ default/1001%192.168.1.2 in the coding.But i am getting the following error: Error: 2008-09-22 10:18:08 [WARNING] mod_sofia.c:1949 sofia_outgoing_channel() Cannot locate registered user [EMAIL PROTECTED] 2008-09-22 10:18:08 [NOTICE] mod_sofia.c:2046 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2008-09-22 10:18:08 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [NO_ROUTE_DESTINATION] 2008-09-22 10:18:08 [WARNING] mod_spidermonkey.c:2933 session_originate() Cannot Create Outgoing Channel! [sofia/default/ 1001%192.168.1.2] 2008-09-22 10:18:08 [ERR] inline:1 mod_spidermonkey() You must call the session.originate method before calling this method! It looks like this phone is not registered. MIke ___ 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] Java script test
thanks for your reply.it is working -- Preetha.A ___ 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
