October 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Recording questions
by Sias Mey 14 Oct '08

14 Oct '08

Hi, Im messing arround with FreeSwitch after the * AMI interface gave me a couple of head and heartaches. Got some call routing today and had loads of fun messing with everyone in the office, of course all of this has to lead somewhere and to that end I'm wondering what the differance is between recording a session, recording a uuid and recording a conferance. are there major CPU or Disk usage variations? I see that recording a uuid does also mix all legs together, does that hold if that uuid is in a conferance as well? or is the conferance record function the way to go if that is the intension? Thanks in advance, Sias

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Questions
by gary 14 Oct '08

14 Oct '08

1) Do I need to reload directory into FS after I added new users in XML file? How? reloadxml? 2) Can I use alias for user? I'd like to register the user with 10 digits DID and also assign 4 digits extension to the same user so people can call each other internally with 4 extensions. What is the best way to do this? I do not want to hard code all these 4 digits extensions into dialplan to map the their DID. 3) Can FS handle large number of sip registrations with multiple(over 100) tenants (context) ? Lets say 10,000 or even 50,000 users. Gary

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ivr and start_dtmf
by Alex Vostrikov 14 Oct '08

14 Oct '08

hi guys, here is ivr i'm trying to use. but i've got a problem that first dtmf digit detected twice. maybe i'm doing something wrong? as a workaround i just added same digit to make it work. <menu name="rostov_aa" greet-long="/opt/freeswitch/sounds/aa/greeting.wav" timeout="10000" inter-digit-timeout="2000" max-failures="3" digit-len="5"> <entry action="menu-exec-app" digits="00" param="transfer 6007 XML default"/> <entry action="menu-exec-app" digits="11" param="transfer 6003 XML default"/> <entry action="menu-exec-app" digits="/^6(60[0123456][0-9]|6070)$/" param="transfer $1 XML default"/> </menu> <extension name="did_2688950"> <condition field="destination_number" expression="^(2688950)$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="start_dtmf" data="true"/> <action application="ivr" data="rostov_aa"/> <action application="set" data="call_timeout=60"/> <action application="set" data="ringback=$${us-ring}"/> <action application="bridge" data="sofia/internal/6002%$${domain}"/> </condition> </extension> 2008-10-14 09:03:24 [DEBUG] switch_ivr_menu.c:755 switch_ivr_menu_stack_xml_build() binding menu action 'menu-exec-app' to '00' 2008-10-14 09:03:24 [DEBUG] switch_ivr_menu.c:755 switch_ivr_menu_stack_xml_build() binding menu action 'menu-exec-app' to '11' 2008-10-14 09:03:24 [DEBUG] switch_ivr_menu.c:755 switch_ivr_menu_stack_xml_build() binding menu action 'menu-exec-app' to '/^6(60[0123456][0-9]|6070)$/' 2008-10-14 09:03:24 [DEBUG] switch_ivr_menu.c:374 switch_ivr_menu_execute() Executing IVR menu rostov_aa 2008-10-14 09:03:24 [DEBUG] switch_ivr_play_say.c:917 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-10-14 09:03:24 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/4992711421(a)172.23.0.1 [BREAK] 2008-10-14 09:03:27 [DEBUG] switch_ivr_async.c:887 inband_dtmf_callback() DTMF DETECTED: [6] 2008-10-14 09:03:27 [DEBUG] switch_ivr_play_say.c:1207 switch_ivr_play_file() done playing file 2008-10-14 09:03:27 [DEBUG] switch_ivr_menu.c:296 play_and_collect() waiting for 4/5 digits t/o 2000 2008-10-14 09:03:27 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Kill sofia/internal/4992711421(a)172.23.0.1 [BREAK] 2008-10-14 09:03:27 [DEBUG] switch_ivr_async.c:887 inband_dtmf_callback() DTMF DETECTED: [6] 2008-10-14 09:03:28 [DEBUG] switch_ivr_async.c:887 inband_dtmf_callback() DTMF DETECTED: [0] 2008-10-14 09:03:28 [DEBUG] switch_ivr_async.c:887 inband_dtmf_callback() DTMF DETECTED: [7] 2008-10-14 09:03:28 [DEBUG] switch_ivr_async.c:887 inband_dtmf_callback() DTMF DETECTED: [0] 2008-10-14 09:03:28 [DEBUG] switch_ivr_menu.c:341 play_and_collect() digits '66070'

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Re: [Freeswitch-users] Originate command
by Jon Bruel 14 Oct '08

14 Oct '08

Brian, I made another test: Originate sofia/gateway/100004444500/201,sofia/gateway/100004444500/205 206 XML internalpreparation This works as it should. So the issue may be related to the user "channel". /Jon

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Re: [Freeswitch-users] Originate command
by Jon Bruel 14 Oct '08

14 Oct '08

Brian, I have now pastebinned the SIP trace as taken from a Snom phone as a response to the command: Originate user/100008888201@10.3.1.24,user/100008888205@10.3.1.24 206 XML internalpreparation I also checked it from the server, and verified that no INVITEs are sent to the address belonging to 100008888205. Further, from the console the line: [DEBUG] switch_ivr_originate.c:700 switch_ivr_originate() variable string 0 = [presence_id=100008888201(a)10.3.1.24] is repeated twice with some 10-20 debug lines in between. To get some way further, I could try to repeat the tests you have done in your lab showing that there is no problem!? /Jon

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seeking a (paid) contractor/firm to develop an application leveraging Freeswitch
by Brian B 14 Oct '08

14 Oct '08

My startup now has budget for a development contract for an application heavily leveraging Freeswitch conferencing. We'll have a web GUI component, an application logic layer, and a layer handling the telephony controls. We could break this project up so that you'd only handle the latter telephony controls (translating some higher level commands into FS commands, and passing events back up the chain). I'm presuming that layer will be in Java. OR, if you/the developer is comfortable writing an Ajax or Flex/Flash GUI as well, one person/firm could handle all of it. Please let me know if you have availability, and send along information about you (or your firm). A portfolio and/or resume is ideal. We're in San Francisco, California, but you could be anywhere. I'm the hiring manager and you can email me directly. Thanks, -Brian freeswitch(a)brian-burt.com

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Reading SIP headers
by Kristian Kielhofner 13 Oct '08

13 Oct '08

Hello everyone, I'm trying to read SIP headers as described here: http://wiki.freeswitch.org/wiki/Channel_Variables#variable_xxxx It sounds like I should be able to grab an arbitrary SIP header and use it in my XML dialplan, right? However I've tried this and it doesn't seem to work. Here's the header I'm attaching: X-conf-pin: 1234 However, reading sip_X-conf-pin doesn't return anything. What am I missing here? Is there any other way to read SIP headers? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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broadvox setup
by Loren Salsgiver 13 Oct '08

13 Oct '08

Can anyone help with connecting freeswith to broadvox. Broadvox provided the following... Allow The Media IPs below and all UDP Ports Up To 65535 209.249.3.58 209.249.3.60 Trunk Number DNS A Record DNS SRV Record IP Addr 1 1000422 dfwnx01ga1.pa.broadvox.net dfwnx01ga1.psrv.broadvox.net 209.249.3.59 As well as a BTN number but no password. In the configuration instructions they stated this is a static connection and all I have to do is point to their server to deliver calls. I did the following... /usr/local/freeswitch/conf/sip_profiles/external I created broadvox.xml, no username and password is required according to the documentation. And none was provided. <include> <gateway name="broadvox"> <param name="realm" value="Broadvox.net"/> <param name="from-domain" value="206.67.234.233"/> <param name="proxy" value="209.249.3.59"/> <param name="expire-seconds" value="3600"/> <param name="register" value="true"/> <param name="retry-seconds" value="3600"/> </gateway> </include> Added to /usr/local/freeswitch/conf/dialplan/default.xml And a dialplan <extension name="To PSTN"> <condition field="fdnis" expression="9541231234"/> <condition field="destination_number" expressions="(.*)"> <action application="bridge" data="sofia/external/$1@209.249.3.59:5060"/> </condition> </extension> No luck. Any assistance would be helpful. If I can get this working the next step would be to interface directly with Verizon using sip. Thanks, Loren

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Re: [Freeswitch-users] Originate command
by Jon Bruel 13 Oct '08

13 Oct '08

Brian, there is a reason for the same address: it's connection to an Asterisk server. My original question remains: Does the originate command work as it should ("originate user/100008888205@10.3.1.24,user/100008888201@10.3.1.24 206 XML internalpreparation")? The problem was that the FS only made a call to one phone, 100008888205, not to both. Maybe I didn't make it clear before that when the call was answered by 100008888205, the phone with extension 206 rung as expected. /Jon

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Creating a Call queue group
by Meftah Tayeb 13 Oct '08

13 Oct '08

hi, please ho to add a number for a call queue ? i want to this number to by a call queue: ringing in all selected users (extentions) if one user accepte the call, other users stop ringing thanks

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