- FreeSWITCH-users - FreeSWITCH Mailing Lists

javascript bridge
by stephen at stephenjc 09 Jun '10

09 Jun '10

I have the following code, and after the bridge the javascript seems to stop. outsession = new Session("{ringback=\'%(2000,4000,440.0,480.0)\',instant_ringback=true,ignore_early_media=false}sofia/gateway/" + providerhash["providername0"] + "/XXXXXXXXXXXX"); bridge(session,outsession); while(outsession.ready()) { console_log("notice","ping"); } I am looking to manage the b leg, should i use execute_on_answer instead? or is there a way to make the code continue after the bridge. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print

4 3

mod_lcr and memcache
by Madovsky 09 Jun '10

09 Jun '10

Is mod_lcr uses memcache for DB request ? Thanks Franck

2 1

Hi
by Zuhair Raza 09 Jun '10

09 Jun '10

Thank you sir it clearified everything. I have one more question to ask, I am unable to compile alsa-driver-1.0.20 on centos 5.5, I googled but could not find any usable info I got this error. In file included from /usr/src/alsa-driver-1.0.20/acore/hwdep.c:1: /usr/src/alsa-driver-1.0.20/include/adriver.h:1779:1: error: unterminated #if /usr/src/alsa-driver-1.0.20/include/adriver.h:1:1: error: unterminated #ifndef make[3]: *** [/usr/src/alsa-driver-1.0.20/acore/hwdep.o] Error 1 make[2]: *** [/usr/src/alsa-driver-1.0.20/acore] Error 2 make[1]: *** [_module_/usr/src/alsa-driver-1.0.20] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-194.3.1.el5-i686' make: *** [compile] Error 2 Then I tried to compile alsar-drive-1.0.21 it was successfully compiled and installed but my skype client does not show dummy hardware. What should i do? On Mon, Jun 7, 2010 at 5:19 PM, Giovanni Maruzzelli <gmaruzz(a)celliax.org>wrote: > Zuhair, > > maybe you need to find a friend with some more unix experience, that > can help you for the first few days. > > You have to: > > 1) compile skypopen_auth as in the wiki ("gcc -Wall -ggdb > skypopen_auth.c -o skypopen_auth -lX11") > 2) login with ssh -X > 3) launch Skype client > 4) launch "skypopen_auth localhost:10.0" > > -giovanni > > On Mon, Jun 7, 2010 at 2:05 PM, Zuhair Raza > <engineerzuhairraza(a)gmail.com> wrote: > > you mean to use this syntax > > gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:101 ?? > > > > When i do echo $DISPLAY it says localhost:10.0 > > and when gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11:$DISPLAY > > it says cannot find -lX11:localhost:10.0 > > > > > > On Mon, Jun 7, 2010 at 4:36 PM, Giovanni Maruzzelli <gmaruzz(a)celliax.org > > > > wrote: > >> > >> On Mon, Jun 7, 2010 at 1:29 PM, Zuhair Raza > >> <engineerzuhairraza(a)gmail.com> wrote: > >> > ok thanks.. but what about my question that skype didnot ask me to to > >> > connect to skype API. and when i do either on ./skypopen_auth :101 or > >> > :0 it > >> > says cannot open X display > >> > >> I told you in previous mail. > >> > >> You need to launch skypopen_auth from the same ssh -X from which you > >> launched the Skype client. And you must give skypopen_auth the correct > >> xserver as an argument. > >> You can use $DISPLAY, or you can check it with "echo $DISPLAY" and > >> then use that value. > >> > >> > >> > > >> > On Mon, Jun 7, 2010 at 3:53 PM, Giovanni Maruzzelli > >> > <gmaruzz(a)celliax.org> > >> > wrote: > >> >> > >> >> On Mon, Jun 7, 2010 at 12:45 PM, Zuhair Raza > >> >> <engineerzuhairraza(a)gmail.com> wrote: > >> >> > Thanks for explanation Sir, One more question, please explain > >> >> > > >> >> > cd /root > >> >> > mount /dev/hda5 /mnt > >> >> > cp /mnt/root/skypeconfig2.tgz ./ > >> >> > tar xzf skypeconfig2.tgz > >> >> > chown -R root.root .Skype > >> >> > > >> >> > According to wiki we haven't created a tgz file before that, but > >> >> > .Skype > >> >> > directory at the server > >> >> > >> >> If you do it with ssh -X (as pre the previous mail), you don't need a > >> >> Skype config directory from another computer. You created that > >> >> directory. > >> >> So, just skip those steps. > >> >> > >> >> -giovanni > >> >> > >> >> > > >> >> > > >> >> > On Mon, Jun 7, 2010 at 12:07 PM, Giovanni Maruzzelli > >> >> > <gmaruzz(a)celliax.org> > >> >> > wrote: > >> >> >> > >> >> >> On Sun, Jun 6, 2010 at 6:52 PM, Zuhair Raza > >> >> >> <engineerzuhairraza(a)gmail.com> wrote: > >> >> >> >> version. First i enter command at my freeswitch box under > >> >> >> >> mod_skypopen/configs > >> >> >> >> > >> >> >> >> gcc -Wall -ggdb skypopen_auth.c -o skypopen_auth -lX11 > >> >> >> >> > >> >> >> >> then I install xauth on server after that i ssh with x > forwarding > >> >> >> >> to > >> >> >> >> my > >> >> >> >> server from an another linux desktop and open skype by typing > >> >> >> >> /usr/bin/skype, it launched skype client at Linux desktop but > it > >> >> >> >> didn't > >> >> >> >> ask > >> >> >> >> for connecting with skypopen api, although it creates ".Skype" > >> >> >> >> directory on > >> >> >> > >> >> >> After launching the skype client from the ssh -X session, you have > >> >> >> to > >> >> >> launch skypopen_auth from the same ssh -X session (giving the X > >> >> >> server > >> >> >> as an argument), eg: "./skypopen_auth $DISPLAY" > >> >> >> > >> >> >> >> my server but when i load mod_skypopen it says could not find > any > >> >> >> >> skype > >> >> >> >> instance and when i typed skypopen_auth it also says no skype > >> >> >> >> instance > >> >> >> >> found > >> >> >> >> on X 0:0. Can anyone tell me where I have mistaken?? > >> >> >> > >> >> >> After having given the auth to the skype client, and closing it so > >> >> >> it > >> >> >> save that auth, you close the ssh -X session, and launch an X > server > >> >> >> and a skype client in the server, using the script in the configs > >> >> >> directory (as explained in the wiki). > >> >> >> > >> >> >> After having launched that script, you load mod_skypopen. > >> >> >> > >> >> >> You MUST edit the script and the skypopen.conf.xml to use your own > >> >> >> values for skype username and password. > >> >> >> > >> >> >> -- > >> >> >> Sincerely, > >> >> >> > >> >> >> Giovanni Maruzzelli > >> >> >> Cell : +39-347-2665618 > >> >> >> > >> >> >> _______________________________________________ > >> >> >> FreeSWITCH-users mailing list > >> >> >> FreeSWITCH-users(a)lists.freeswitch.org > >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> > >> >> >> > >> >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> http://www.freeswitch.org > >> >> > > >> >> > > >> >> > > >> >> > -- > >> >> > Regards, > >> >> > Zuhair Raza > >> >> > > >> >> > > >> >> > _______________________________________________ > >> >> > FreeSWITCH-users mailing list > >> >> > FreeSWITCH-users(a)lists.freeswitch.org > >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > > >> >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> > http://www.freeswitch.org > >> >> > > >> >> > > >> >> > >> >> > >> >> > >> >> -- > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> Cell : +39-347-2665618 > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users(a)lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > -- > >> > Regards, > >> > Zuhair Raza > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users(a)lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> > >> > >> -- > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Regards, > > Zuhair Raza > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Zuhair Raza -- Regards, Zuhair Raza

2 2

CHANNEL_PARK event not triggered anymore from event socket?
by Peter Olsson 09 Jun '10

09 Jun '10

I'm not filing this as a bug right now, since I don't know if this was an intended change. I used to trigger on the CHANNEL_PARK event, but it doesn't seem that this event is fired anymore. Is this change intended? If not, I will submit a jira report with full logs showing this issue. /Peter

2 3

Need to stop more than one voice file using break application
by Thangappan.M 08 Jun '10

08 Jun '10

Dear all, I am in the process of developing IVR using FreeSWITCH. For that I am being used outbound ESL in async mode. In my design, usually for one menu it might be more than one voice files. So using playback_delimiter, play back all the voice file in single instance using playback application. I've tried to stop the play back using "break <uuid> all" API. But it only break the only one voice file not the whole application(playback). Consider that I am playback four voice files. When the first voice file is getting playback, using " break <uuid> all" t stop the playback . It only stopped the second voice and continued to playback the third voice file and followed by fourth one. My need is to break the playback application which may have any number of voice files in async mode. Thanks in advance. -- Regards, Thangappan.M

3 6

Out-Going Call Transfer Question
by Michael Gende 08 Jun '10

08 Jun '10

Please forgive, I sent this follow-up a few days ago but think I accidentally deleted the response. Again, I've a PFsense FS install that works great. But, and this is probably something I did, I can't transfer a call session originating here to anywhere but a conference room (i.e., I can't transfer a call I made to another extension. It hangs up the call). For example, I just called to my cell phone from FS. Then, having answered it, transfer the call to my colleague at ext 1012. I'm extension 1013. The transfer hangs up the call on both ends. Console output follows: *2010-6-03 12:51:42:950492 [ERR] Switch_ivr_originate.c: 1494 Cannot Create outgoing channel of type [user] Cause: [SUBSCRIBER ABSENT] 2010-6-03 12:51:42:950492 [INFO] mode_dtools.c: 2091 Originate Failed. Cause: SUBSCRIBER_ABSENT* * 2010-06-03 12:51:43.960389 [WARNING] mod_voicemail.c:2931 Can't Find User [1012(a)wan.ip.address (should be LAN ip typically, no?)] * > Also, is this a plain FS install with the default dialplan? If not then > include your relevant dialplan entries. Our install is pretty well documented in the following: http://wiki.freeswitch.org/wiki/Multi_home_tutorial The setup described here is what we have. > Lastly, are you on the most recent hit HEAD version of FS? > Nope. This is FS on PFSense, the Beta 0.9.7.26 version. No other issues, however, despite the Beta moniker. Is the older version the issue? Sorry for the re-post. Any comments most appreciated. Mike G.

1 0

Multi-user on-demand conference
by David Swardstrom 08 Jun '10

08 Jun '10

The JavaScript examples include a sample similar to this. With a little bit of work, this should be possible to develop. Look at http://wiki.freeswitch.org/wiki/Javascript_Examples One difference (should not be significant) is that as you add each participant, they would be added to the conference. If you use the wait-mod flag in the conference, the participants would hear music-on-hold. When the originator has added all of the participants, then add the originator as the Moderator. This action adds the originator/moderator to the conference, the MOH ends, and everyone can talk to each other. Regards,David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

1 0

Messaging: Content-Type: application/octet-stream changed to text/html.
by Peter P GMX 08 Jun '10

08 Jun '10

Hello, we have a SIP based Videoconferencing solution which we would like to integrate in Freeswitch. Registering the endpoints works, however all other operations which are done via messaging do not work as expected, as Freeswitch changes the Content-Type, see below. How can we tell Freeswitch not to change the Content-Type? Best regards Peter Original message from UA to Freeswitch with "Content-Type: application/octet-stream.": U 2010/06/01 18:21:57.056468 192.168.178.145:5530 -> 192.168.178.220:5060 MESSAGE sip:835351@my.domain SIP/2.0. From: Hans <sip:835352@my.domain>;tag=de58b276076cdf119f8bcf0221b28bff. To: sip:835351@my.domain. Via: SIP/2.0/UDP 192.168.178.145:5530;branch=z9hG4bKae56b276076cdf119f8bcf0221b28bff;uas-addr=192.168.178.220. CSeq: 3 MESSAGE. Call-ID: 645ab276-076c-df11-824f-cf0221b28bff. Contact: "Hans"<sip:835352@192.168.178.145:5530>. User-Agent: BRAVIS/0.0.0.27.4675 (Linux 2.6.32-22-generic; generic; Ubuntu 10.04 LTS; i686; de; 8). Content-Type: application/octet-stream. Content-Length: 231. . BRVSDRABAAEAAACeAAAAGfLjfQQAAAAvrRdulaIBAAAAMQEA/wAASwUAAABzcmZseP////+n pgi9D7MvrRduD7MBAAAANAEA/QAASwUAAABzcmZseP////+npgi9D7Onpgi9D7MBAAAAMwEA BVHDfgQAAABob3N0/////5WmD7d9pQQUp3cPswEAAAAyAQADUcN+BAAAAGhvc3QA////laYP t32lAgAAAA==. Modified message from Freeswitch to another UA with "Content-Type: text/html": U 2010/06/01 18:21:57.061965 192.168.178.220:5060 -> 192.168.178.122:5530 MESSAGE sip:835351@192.168.178.122:5530;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.178.220;rport;branch=z9hG4bKaHZUBrU7S2BXQ. Max-Forwards: 70. From: Hans <sip:835352@my.domain>;tag=de58b276076cdf119f8bcf0221b28bff. To: "Peter "<sip:835351@192.168.178.122:5530;transport=udp>. Call-ID: a3bd6fee-e83c-122d-1ca4-080027e51f59. CSeq: 131586180 MESSAGE. Contact:<sip:mod_sofia@192.168.178.220:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Content-Type: text/html. Content-Length: 233. . BRVSDRABAAEAAACeAAAAGfLjfQQAAAAvrRdulaIBAAAAMQEA/wAASwUAAABzcmZseP////+n pgi9D7MvrRduD7MBAAAANAEA/QAASwUAAABzcmZseP////+npgi9D7Onpgi9D7MBAAAAMwEA BVHDfgQAAABob3N0/////5WmD7d9pQQUp3cPswEAAAAyAQADUcN+BAAAAGhvc3QA////laYP t32lAgAAAA==.

2 1

Multi-user on-demand conference
by Yehavi Bourvine 08 Jun '10

08 Jun '10

Hello, Our users are used to create conference in the following way: - Call the first participant. - press "conference" which puts the other participant(s) on hold. - Call the new participant and talk to him/her. - press "conference" again to join everyone. - Repeat the above process for additional participants. They would like to do the same with Freeswitch also, and for large number of users (more than 3-4 which the phones can do themselves using the above procedure). Has anyone done this and can send an example? I would like to define one of the keys of the phone to send some * code which would behave like the above "conference" button and instructs Freeswitch to handle the conference. Thanks! __Yehavi:

1 0

LUA MySQL connection pool
by Tony Tin 08 Jun '10

08 Jun '10

Hi, I'm using Lua script and MySQL to do dynamic dialplan. I'm wondering whether I need to create a connection pool for all calls or I just create a new connection for every call. How can I create the connection pool in Lua? Thanks. Tony

1 0