December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Inbound calls
by whudson05 26 Jan '09

26 Jan '09

Hi, I'm experiencing a very specific problem to do with inbound calls through a gateway. When either end of the phone call hangs up then the other end will remain connected as if still in the conversation. If anyone has any idea why this would occur it would be greatly appreciated. Thanks -- View this message in context: http://www.nabble.com/Inbound-calls-tp18694165p18694165.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 4

Conference and socket outbound
by Dennis 23 Jan '09

23 Jan '09

hi, now, that everything is working, what i wanted to do with freeswitch, i want to look a little bit more left and right. one interesting thing to "play" arround with, is conferencing. is it possible to do conference calls, when using socket outbound and php? where could i start? ok, of course with "mod conference" in the wiki. but i wonder, how i can put a caller in a conference room and how to set the properties (what dtmf does what and which soundfiles are played when) dynamically. let's say i get a socket inbound call and i answer it. then i play a soundfile, which asks to enter the pin... how can i move the inbound call or originated calls into a conference room? it seems, as if one can do a lot with api in the conference room. but i wonder what i have to do in the beginning, to put people (specific uuid's) into a conference room. thanks, dennis

3 3

Channel variable 'call_timeout'.
by henkoegema 16 Jan '09

16 Jan '09

I use following diaplan for extension 2005: --------------------------------------------------------- <extension name="2005"> <condition field="destination_number" expression="^2005$"> <action application="set" data=”call_timeout=10"/> <action application="set" data="continue_on_fail=USER_BUSY,NO_ANSWER"/> <action application="bridge" data="sofia/internal/2005%$${domain}"/> <action application="voicemail" data="default $${domain} 2005"/> </condition> </extension> What I can't get to work is the "call_timeout" variable. It is set to value 10 at the moment. So I was thinking, that if exten. 2005 is busy or doesn't answer, the call would be transferred to the voicemail after 10 seconds. It is actually transferred after 66 (?) seconds. Changing the value of call_timeout doesn't seem to have any effect. :confused: -- View this message in context: http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20677… Sent from the Freeswitch-users mailing list archive at Nabble.com.

7 16

mod_skypiax inching forward
by Giovanni Maruzzelli 09 Jan '09

09 Jan '09

Hi FreeSWITCHers! mod_skypiax, the Skype compatible endpoint, is slowly inching toward release :-) When the demo is online (will go on and off for development), you can test it (so helping finding bugs) by calling with Skype the Skype Names: skypiax20, skypiax19, skypiax18, ...., skypiax1 Happy New Year !!! Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039

2 1

close channels properly
by Carole O. 07 Jan '09

07 Jan '09

Hello, I use the following code to call VoIP speakers and make an announcement: <extension name="security_announcement"> <condition field="destination_number" expression="^9001$"> <action application="set" data="conference_auto_outcall_caller_id_name=security_announcement"/> <action application="set" data="conference_auto_outcall_id_number=9001"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_announce=announcement/announcement_security_delay.wav"/> <action application="conference_set_auto_outcall" data="user/1021@$${domain}"/> <action application="conference_set_auto_outcall" data="user/1022@$${domain}"/> <action application="conference" data="conf_speaker@profilespeaker_announcement"/> </condition> </extension> 1021 and 1022 are the speakers. At the end of the announcement, since there is no noise anymore, the speakers stop listening but they do not send any messages to tell Freeswitch it can close the opened channels. The channels are closed only after a timeout of 5 minutes. Does anybody know how I could force freeswitch to close all the channels after the announcement? I have seen there is the application sched_hangup but when I used it it only closes the channel to the caller and not the other ones. Thanks a lot for your help and Happy Holidays! Carole -- View this message in context: http://www.nabble.com/close-channels-properly-tp21127913p21127913.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

4 8

busy tone detection
by Gopal krishnan 05 Jan '09

05 Jan '09

Hi, Is it possible to have busy tone detection thru console dialing? -- Thank you with regards, Gopal,

6 23

Conference
by Baskar 03 Jan '09

03 Jan '09

Hi, *Just i want to call the 1002==>9841799874 and i want to call another no 9840544078 and add in to the conference room * *First method i tried :* api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874(a)172.20.191.228) ====>1002 call 9841799874 api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Both caller and extension will be transfer into conference room Then i call the another caller api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9840544078(a)172.20.191.228)====> 1002 call another caller 9840544078 api uuid_transfer c564f668-6778-4e78-85f6-a78c9c67cfa4 -bleg 3001 ====>Transfer only the caller 9840544078 to conference room 3001 while calling the second no 9840544078 the first line get hold after the transfer of bleg they all come in to the conference room but the first line caller is in hold so we cant able to unhold the call through api we have to manually go the soft phone and unhold the call . but i need the unhold should be done in api what is the solution . *i have tried in another method also * api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874(a)172.20.191.228)====>1002 call 9841799874 api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Transfer only the caller 9841799874 to conference room 3001 api uuid_kill 04120c08-19e7-4881-8873-0834a11524ff before i hangup i have transfer both the caller in to conference room so when i hangup the extension why my extension got hangup Can any one suggest me how can i try the conference are else correct where i am wrong. -- Warm Regards, N.Baskar

4 6

uuid_playback
by Peter P GMX 01 Jan '09

01 Jan '09

As I see on the Wiki page uuid_playback seems to be implemented, however it doesn't work on the console or via event_socket. Also in the code I could not find it (svn 10438). So for now I use uuid_brodcast to play announcements to one or both parties. Question: What is the status of uuid_playback? Best regards Peter

3 4

New
by FERNANDO VILLARROEL 01 Jan '09

01 Jan '09

Dear All, I am new on this list, i am chilean. I come from Asterisk and my company is dedicated to wholesale, i am provider for chile mobile. We will need implement FreeSwictch before Asterisk for Retail and Wholesale How i will can starting, please recomended any how to or pdf for beginning. Is possible administrator FS from any GUI, but no from Centos Image FS,I like Debian. Other questions, FreeSwitch is more complex that Asterisk for begining? Thanks everyone and excuse me English. Fernando

4 4

Looking for a specific bug in jira - was Sonus and RFC 2833 related
by Kristian Kielhofner 31 Dec '08

31 Dec '08

Hey everyone, A few days ago I was reading about a bug with Sonus gear and FreeSwitch. I remember it because Tony implemented a workaround the buggy Sonus gear based on the SDP, all while adding a snarky (yet hilarious) log message. You guessed it, I'm having 2833 timestamp problems with Sonus... I can't find this issue for the life of me in Jira. Does anyone know where it is? It was fixed in rev 10744 with the commit message "sonus, sonus, sonus sonus is a four letter word". Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

4 8