January 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

network scenario doubts
by Michal Kopacki 28 Jan '11

28 Jan '11

Hello, This is my first post to this list, so hello everyone. I'm at the very beginning of freeswitch journey and i have a problem with fit FS to my network scenario. OS: fedora 13 (x86_64) FS: 1.0.7 (compiled from sources), default config Desired scenario: softphone -> mydomain.com (with ip 193.59.72.xx) -> my isp network -> 10.25.48.xx on eth1 -> FS -> 192.168.0.1 on eth0 -> softphone And now, I'm able to connect to FS from my local network, but I'm not able to connect from outside (neither domain nor ip). In internal.xml I set internal ip of server and i wanted to set external ip in external.xml, but there is a question: which one ? In case of 193.59.72.xx external profile didnt' start and with 10.25.48.xx outside softphone didn't register. I checked with netstat and realized that port 5060 is bind to internal nic only and 5080 to external nic (10.25.48.xx) and I have no idea what next. Is it even possible to work with such network scenario ? I would be grateful for pointing me to right direction or maybe propose different approach. -- Best regards, Michal

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TBCT 2bchannel transfer
by Holger Esser 28 Jan '11

28 Jan '11

Hi, does anybody have a detailed description of how to implement the TBCT transfer method with Sangoma cards? I was looking at the att_xfer but I am not sure if that would cover TBCT. Any suggestions would be greatly appreciated. Holger

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nibblebill and ODBC
by Madovsky 28 Jan '11

28 Jan '11

I don't know yet if it's related to ODBC, but if I use nibblebill pause, (works first time) hangup and call the same again so the dialplan is held at nibblebill pause infinetly. I have to restart FS (on console I can see nibblebill is waiting some reference or process) ? thanks

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Number mapping (DID like)
by Budi wibowo 28 Jan '11

28 Jan '11

Hi I want to do B# (called) rewriting, for example >From user 1000 call to 1001, and freeswitch rewrite 1001 to other number (pstn). Is it doable with freeswitch? Regards Budi

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supported say languages
by Holger Esser 28 Jan '11

28 Jan '11

Hi, is there a list of all the supported say (mod_say_xx) modules? I am looking at my build and so far I am seeing Russian and English. Are there more? Kind regards, Holger

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Question about IVR Menu
by DJB International 28 Jan '11

28 Jan '11

In IVR menu, I have: <entry action="menu-play-sound" digits="2" param="voicemail/company_directory.wav"/> to play company directory when the caller press 2, but the caller have to wait until the wave file finished playing to the end of the file before he can enter any digits . Is there a way to enter digits (extension number) at any time without waiting for the wave file to finish playing? Thank you.

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mod_dingaling answering prematurely
by Jon Young 28 Jan '11

28 Jan '11

I just got mod_dingaling working for outbound and inbound calls. However, on inbound calls I have to answer it quickly on my SIP phone. When I do I catch the call screening announcement from google to press a "1" to answer. I put the "execute_on_answer=send_dtmf 1" as recommended in the Wiki which eliminates the need to press one. It seems that FS is answering the call and google starts playing the screening. If I wait too long to answer I miss the announcement. I am routing the call to a specific extension for testing. The dialplan is using the extension definitions provided in the default configuration. Where should I start looking to allow the phones to ring longer and not prematurely answer the inbound call? Thanks, Jon

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question about outbound socket async
by Seven Du 28 Jan '11

28 Jan '11

Hi all, I'm using mod_erlang_event and I think it's equivalent to event_socket async full. When I run sendmsg playback wav1 and then playback wav2, wav2 will stop wav1 and when wav2 is done, wav1 continues by default. Is there a way to automatically break wav1 when play wav2 ? I could send break to wav1 but because everything is async, sometimes break also breaks wav2 even with event-lock = true which made async mode a sort of pain. It seems like there is a var can control this but I cannot recall. Any help is appreciated. Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Choppy VM
by Frank Park 27 Jan '11

27 Jan '11

Hey guys, I've been trying to troubleshoot this, but haven't been able to figure it out... I am currently experiencing choppy voicemail recordings on my freeswitch. The calls are coming in using SIP trunk and the call quality using the same trunk has no problem, but when I listen back the vm, that can be a different story. This is running on a dedicate server and not only a virtual instance. I am currently not using mod_shout, so this is being saved as a wav file. I've monitored CPU and memory usage while this happens, and because they are so low, I don't think resource exhaustion is a factor. Any direction as to what I should try to troubleshoot what's wrong? I appreciate any input. Thank you, Frank -- ----=======================---- Frank Park Telonium Communications, LLC frank(a)telonium.com http://www.telonium.com Follow Us on Twitter: @GetTelonium 404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----

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conference with did example
by Miguel Miranda 27 Jan '11

27 Jan '11

Hi, i have searched the arvhices and could not find a simple example on how to configure a DID for mod_conference, i mean you dial an DID, an ivr responds "please enter you conference number and press puond key, etc", and if the conference room was configured with pin it asks "please enter ´pin number, etc". as i undestend, in this example from wiki: <extension name="test_bridging_conference"> <condition field="destination_number" expression="^(3000)$"> <action application="answer"/> <action application="playback" data="connecting_your_call.wav"/> <action application="set" data="ringback=${us-ring}"/> <action application="conference" data="bridge:$1-${domain_name}@default:user/1000@${domain_name}"/> </condition> </extension> i need a gateway and if you receive the DID 3000 you join the conference right? What i want is a single did for all the conferences. regards

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