July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

One Week Until ClueCon MMX!
by Michael Collins 27 Jul '10

27 Jul '10

Everything is gearing up for another great ClueCon event! We have speakers, sponsors, giveaways and even a few surprises. Rumor has it that there might be a few copies of the new FreeSWITCH book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tel…>to give away... If you have not already registered for ClueCon then please do so right away. Use the online form <http://cluecon.com/> or call us at 877.742.CLUE(2583). I just spoke with the Trump and as of right now they are still accepting reservations for the discounted rate of $225 per night! There are still a small number of rooms available so definitely get your reservation today. Thanks to all of our speakers, sponsors, and attendees for always making ClueCon the best conference of the year! The ClueCon Team http://www.cluecon.com 877.742.CLUE (2583)

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Issue with delayed hangup when originating from JS
by Sias Mey 27 Jul '10

27 Jul '10

Hi, I am working on porting an outbound CC app from asterisk to FS. method: I originate a call to the agents phone and if they answer I bridge/originate and then bridge/connect to a conferance and transfer to an outbound line. In all of those scenarios if I use JS to do two seperate originates I have the same issue. problem: If the agent hangs up his phone before a connection on the second originate both lines stay open untill something happens to the second leg. If there is an originate timeout on it and it isent answered then both legs with hangup as soon as the second leg terminates as well. If the client picks up the phone then both will teminate too, this of course leads to a really bad "ring ring, hello?, click" situation with the agent not even aware that it is happening. If I use a single originate directly to a bridge then both legs terminate as soon as the agent/first leg is hung up, with both api_hanguphooks running as expected. It seems to only be an issue if I originate to park() and then connect the two via JS or if I originate directly to a JS file that then generates and bridges the other session. I have managed to work arround it with a fairly rediculous originate api call moving some lcr and balancing code into my frontend rather than a FS JS script. But this means my application has to be aware of the number of calls per lcr route and all sorts of things that I would have prefered to handle in FS directly. I dont know if this is a design feature to alleviate the post hangup code issues that I had in asterisk, or if it is just the way I am implementing it that is messing me arround. So if one of you guys that know the internals really well can help me out with a reason or workarround for this behaviour I would greatly appreciate it. Again there is no urgency to this anymore since my current implementation works fine. But I would like to understand the underlying causes of this behaviour if that is possible to explain without a "go read the code" :-D. Thank you very much, Sias

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Re: [Freeswitch-users] mod_spandsp.c - compile error with "make current"
by Thomas Mueller 27 Jul '10

27 Jul '10

On 27.07.2010 14:40, Peter Olsson wrote: > Try with ./bootstrap.sh and ./configure, then make current again. > indeed, this fixed the problem. thanks Peter! - Thomas > ----- Reply message ----- > Från: "Thomas Mueller"<thomas(a)chaschperli.ch> > Datum: tis, jul 27, 2010 14:22 > Rubrik: [Freeswitch-users] mod_spandsp.c - compile error with "make current" > Till: "freeswitch-users(a)lists.freeswitch.org"<freeswitch-users(a)lists.freeswitch.org> > > hi > > since a few days "make current" fails to compile on my Debian 5.0 "Lenny" > machine: > > Compiling mod_spandsp.c ... > In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp/ > expose.h:81, > from /usr/local/src/freeswitch/libs/spandsp/src/ > spandsp.h:127, > from mod_spandsp.h:46, > from mod_spandsp.c:37: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:106: > error: expected specifier-qualifier-list before ‘t4_t6_decode_state_t’ > make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/ > applications/mod_spandsp' > make[4]: *** [mod_spandsp-install] Error 1 > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > > > > - Thomas > > > _______________________________________________ > 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 > > !DSPAM:4c4ecf8a32932003815233! >

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Outgoing call from SIP phone
by Raymond Chan 27 Jul '10

27 Jul '10

Hi all, I am configuring Freeswitch (Windows) and have SIP phone registered. The Freeswitch is connecting a Cisco router with VWIC T1 connection. However, I cannot forward the outbound from SIP phone to Cisco. Did I configure wrong? I tried using gateway, but it always "invalid gateway". What should I put in dialplan? sofia status Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia@203.194.134.91:5080 RUNNING (0) external profile sip:mod_sofia@203.194.134.91:5060 RUNNING (0) 192.168.100.2 alias internal ALIASED ============================================================================ ===================== Dialplan for outgoing call. <extension name="Outbound_call_from PHI"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bridge" data="sip:98403073@192.168.100.51"/> </condition> </extension> Thanks Raymond

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FreeSwitch Firewall Configuration
by Jigar Thakkar 27 Jul '10

27 Jul '10

Hi List, My Freeswitch server is behind firewall (Cyberoam), and want to connect it from outside network. Please help me to configure the same. I have already opened 5060 and 5080 ports for SIP. I am able to register to FreeSwitch, call is established but I can't hear voice. Can you guys please help me with this? thanks & Regards, Jigar.

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Voicemail IVR tree configuration
by Maciej Bylica 27 Jul '10

27 Jul '10

Hello, I have working voicemail module, but i cannot find a way to turn off incoming messages after the voice message is recorded. In other words to omit operator messages like "press 1 to listen ....., press 2 to save, ... (after a few secs of silence). Where can i find a voicemail ivr tree? Thanks, Maciej

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Fwd: How to invoke a script in freeswitch similar to AGI in asterisk
by Zuhair Raza 27 Jul '10

27 Jul '10

Dear Anthony Sir, I used esl and did the ivr example but it was breaking socket connection when remain idle for few hours even though i was connected to localhost. Now i am using mod_xml_curl on Mr. Sergey's advice. On Mon, Jul 26, 2010 at 11:00 PM, Anthony Minessale < anthony.minessale(a)gmail.com> wrote: > look on our wiki for ESL > > > On Mon, Jul 26, 2010 at 12:02 AM, Janu Mukherjee <janu.mukhi(a)gmail.com>wrote: > >> >> Hi, >> >> I am very new to freeswitch. I need to execute an application. On asterisk >> the dial plan looks as follows >> >> exten => 1500,1,AGI(agi://192.168.0.131/hello.agi). >> >> I need something similar to this. How do i do this from freeswitch. Please >> help me in this regard. >> >> Thanks & Regards, >> Jahnavi. >> >> _______________________________________________ >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> > pstn:+19193869900 > > _______________________________________________ > 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

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How to handle inbound calls?
by ovvenkat 27 Jul '10

27 Jul '10

Hi to all, I can successfully handle outbound calls. Can you any one tell me, how to handle inbound calls? Regards Venkatesan OV.

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How to invoke a script in freeswitch similar to AGI in asterisk
by Janu Mukherjee 27 Jul '10

27 Jul '10

Hi, I am very new to freeswitch. I need to execute an application. On asterisk the dial plan looks as follows exten => 1500,1,AGI(agi://192.168.0.131/hello.agi). I need something similar to this. How do i do this from freeswitch. Please help me in this regard. Thanks & Regards, Jahnavi.

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how to make a call from freeswitch
by Abubacker siddiq 26 Jul '10

26 Jul '10

Dear all, I dont have any hard phones and softphones , but I have installed freeswitch in my machine and I have headset with speaker in that machine. I want to make an outbound call to the some one who is in the same environment and he is having a softphone ( twinkle ). Is this possible , if yes how to do this ? Thanks in Advance ! -- BEST REGARDS N.ABUBACKER SOFTWARE ENGINEER BK SYSTEMS (P) LTD CHENNAI-23

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