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Background music in conversation
by Ashutosh 12 Jul '08

12 Jul '08

Hi, The scnario required is as follows: A calls B using FS, and a music should be played in the background while they are talking. I have searched whole wiki and fs docs and mod_dptools but unable to find a pointer so far. Even in asterisk, this has no plain solutions, other than to bring both the parties in a meetme room and starting moh for a third party on local/ channel. Anyone help me out or any links ? Best Regards, ashutosh

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FreeSWITCH and Amazon EC2 port problem
by Diego Viola 12 Jul '08

12 Jul '08

Hi I have FreeSWITCH 1.0.0 installed on Amazon EC2, today was working very well... but after I re-installed FS I'm getting some problem with ports. I did followed this guide and opened the required ports: http://wiki.freeswitch.org/wiki/Amazon_ec2 The problem is when I run FS, I see that the normal ports are opened: freeswitch@Etch> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia@67.202.9.134:5080 RUNNING (0) nat profile sip:mod_sofia@67.202.9.134:5070 RUNNING (0) default alias internal ALIASED internal profile sip:mod_sofia@10.251.203.143:5060 RUNNING (0) ec2-67-202-9-134.compute-1.amazonaws.com alias internal ALIASED outbound alias external ALIASED ================================================================================================= 3 profiles 3 aliases But when I do a netstat -a, I see that 5060 is not opened... there is a 5070 port opened though, but my softphone (X-Lite) doesn't register... what could be wrong? Thanks, Diego

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freeswitch interop with Altigen PBX
by Scott Barr 12 Jul '08

12 Jul '08

Is it possible for freeswitch to communicate directly with an Altigen's PRI interface w/hardware or would freeswitch be able to send transfers as SIP INFO + flash as this is how Altigen support says the Altigen PBX handles transfer requests via SIP. Also, does anyone have any idea where I can find documentation os the specifics of using freeswitch as an SBC? Thanks. -Scott

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Unvoluntary hangup, but OK after restart
by Ivan C Myrvold 11 Jul '08

11 Jul '08

My FreeSWITCH is behind nat, and I have calls coming in from internet on port 5060. This works great after I got the IP range set in acl.conf.xml. But the calls get hung up after 40-50 seconds into the call for no reason at all. But after I restart FreeSWITCH, this annoying behaviour doesn't happen. I traced both cases, when the call is hung up here: http://pastebin.freeswitch.org/4784 It looks like FreeSWITCH is resending SIP/2.0 200 OK for some reason, until the other end gives up and sends a BYE sip:Unknown@81.201.82.28 SIP/2.0 But after a restart, FreeSWITCH behaves much better, and the call goes on until one of the parties hang up normally: http://pastebin.freeswitch.org/4785 I am not sure if this is a bug, or a misconfiguration of FreeSWITCH FreeSWITCH Version 1.0.trunk (8900) Ivan

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breaking the ice with IVR
by Julien Chavanton 11 Jul '08

11 Jul '08

I am making test in order to experiment IVR application with Freeswitch, I have spent a few hours trying, reading and searching the wiki. I am starting by looking for XML or Perl scripting. - How do we call a perl script from a dialplan like "default_context.xml" <extension name="perl_test"> <condition field="destination_number" expression="5000"> <action application="perl" data="/tmp/xml.pl"/> </condition> </extension> - I have tried to test the IVR application "conf/lang/en/demo/demo-ivr.xml" <extension name="ivr_demo"> <condition field="destination_number" expression="5000"> <action application="ivr" data="demo_ivr"/> </condition> </extension> error: 2008-07-11 10:04:18 [ERR] mod_dptools.c:1147 ivr_application_function() Unable to find menu Do I have to put the xml ivr file somewhere ? Thanks, Julien

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Compile and install mod_cdr
by Rodrigo P. Telles 11 Jul '08

11 Jul '08

Hi Folks, I'm trying to compile and install mod_cdr following the wiki page http://wiki.freeswitch.org/wiki/Mod_cdr but it seems obsolete and broken. BTW I try do add "event_handlers/mod_cdr" on modules.conf and it builds fine. But when I try to run freeswitch it complains with: 2008-05-09 11:35:03 [CRIT] switch_loadable_module.c:727 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_cdr.so **/usr/local/freeswitch/mod/mod_cdr.so: undefined symbol: _ZN12CDRContainer7add_cdrEP19switch_core_session** Does some one here build mod_cdr sucessfully? If so, please share the steps with us. Thanks in advance, Rodrigo Telles

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Directed call pickup
by whudson05 11 Jul '08

11 Jul '08

Hi, I've been looking at using FreeSwitch for the last few days as a PBX. One of the features I'm quite keen on having is directed call pickup. In other words answering an extension that is currently ringing by dialing a code and then the extension number. So far I don't seem to have come across anything about this. Is it possible currently possible or would it require writing a new module? Thanks -- View this message in context: http://www.nabble.com/Directed-call-pickup-tp18385751p18385751.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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bridge and playback
by Mihail Krivushin 11 Jul '08

11 Jul '08

I want to realise next scheme: 1. try call (bridge) to local extensions 2. if no one answer in 20s playback wav with attention to wait 3. bridge to mobile phone I dont understand - I must set <action application="set" data="hangup_after_bridge=false"/>? If I set this and leg b hangup (dont answer or normal clearing) playback will execute? How I can do 2&3 only on NO_ANSWER cause? -- С уважением, Кривушин Михаил Евгеньевич Системный администратор ЗАО "СОЮЗ Т" тел. +7 913 865 7 866 jid: mkrivushin(a)souzt.com icq: 218 744 127

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silence, non-silence or tone detection
by Julien Chavanton 10 Jul '08

10 Jul '08

Hi, According to what I have found in the wiki, these features are still not available in Freeswitch ?

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IVR: dynamic call transfer/bridge
by Birgit Arkesteijn 10 Jul '08

10 Jul '08

Hi, I'm looking at the various ways to implement IVR in FreeSWITCH. My dialplan is generated dynamically, using xml_curl. What I'd like to achieve is: - user rings a number - FreeSWITCH detects the outing number and asks "You've reached XXXX in YYYY. If you want to continue press 1. Alternatively this call will terminate." When the user presses '1', the call has to be transferred to a dynamic number. Can I (and if so, how do I) pass the (dynamic) extension (actually bridge to gateway) from my dialplan to the IVR? ivr.conf.xml seems easy to use. Can I use dialplan variables in ivr.conf.xml, or do I need to use lua or js? I know I can use <action application="set" data="to_be_transferred=XXX"/> Would I be able to use this new variable in ivr.conf.xml or any of the other IVR tools? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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