August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dropping calls after 40 secs
by Brett Wilson 16 Aug '12

16 Aug '12

hello, I am having issues with calls being dropped after about 40 secs. I just compiled and installed the latest fs. Here is log output: pastebin.com/abJJfLUi ******************************************* Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Cell: 973.650.7189 Email: brett(a)launch3.net www.Launch3.net www.Launch3telecom.com *******************************************

3 4

Profiling FreeSWITCH on Centos
by Ken Rice 16 Aug '12

16 Aug '12

Hey Guys, Help me remember. I was speaking with one of you at ClueCon about profiling FreeSWITCH on Centos5 and Centos6 so we can compare whats going on. I can not remember who that was! If it was you, please contact me on IRC (I¹m SwK on #freeswitch on freenode), or Email me off list Thanks Ken

4 6

Any good Canadian SIP carriers
by Phil Kim 16 Aug '12

16 Aug '12

Hello all, I am looking for a good SIP carrier (provider) in Canada. We already have a carrier in U.S. but they do not offer good rates for calls coming over from Canada. Any inputs will be appreciated. Thank you. Best regards, Phil H. Kim The information contained in this e-mail message, and any attachment thereto, is the property of Valiant and is confidential and may not be disclosed without our express permission. If you are not the intended recipient or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that you have received this message in error and that any review, dissemination, distribution or copying of this message, or any attachment thereto, in whole or in part, is strictly prohibited. If you have received this message in error, please immediately notify us by telephone, fax or e-mail and delete the message and all of its attachments. Thank you. Every effort is made to keep our network free from viruses. You should, however, review this e-mail message, as well as any attachment thereto, for viruses. We take no responsibility and have no liability for any computer virus which may be transferred via this e-mail message.

9 8

mod_event_zmq thread safety
by Alex Massover 16 Aug '12

16 Aug '12

Hello, We're using mod_event_zmq module (slightly modified, we use different socket type and IPC instead of TCP), and experience crashes. As far as I understood the module it creates ZMQ context and all event sent by using same context. Events callback is called from different threads, isn't it? And can be a situation when different threads access send() api on this context simultaneously. ZMQ guide strongly prohibits doing that: " You MUST NOT share ØMQ sockets between threads." And when I do a mutex around send() there're no more crashes (at least today). Am I missing something or mod_event_zmq is potentially dangerous? -- Best Regards, Alex Massover

2 2

Windows 8 Metro / Windows Phone 8 Freeswitch SIP Client Challenge
by Mitch Capper 16 Aug '12

16 Aug '12

A few users have asked about using freeswitch as a softphone on Windows 8 metro (or Windows 8 UI or WTF they call it now) or Windows Phone 8. If anyone is able to put together a proof of concept freeswitch working with portaudio (or equiv) at the level they can place a test call and talk (the number and what have you can be hard coded in I just need to see it actually work on these platforms) I will be happy to write a softphone for either or both those platforms. ~Mitch

1 0

caller_id_number
by Boris Kovalenko 16 Aug '12

16 Aug '12

Hello! When the call is incoming for my network I want CID is translated by my rules. So I translate caller_id_number and set effective_caller_id_number. But there is a little problem - when the call arrived to a analog phone (for example Panasonic KTX-TCD540) I see two number on display - original caller_id_number and effective_caller_id_number. Is there a way to not show caller_id_number in CID? Or may I modify caller_id_number instead of setting effective_caller_id_number? -- Regards, Boris

9 15

uuid_broadcast and fifo
by Lynn Nielson 16 Aug '12

16 Aug '12

Is it possible to do a uuid_broadcast into a fifo parked call? At the command line it tells me the Message is Sent but I do not hear it on the call. The equivalent commands I'm using from fs_cli are as follows: originate sofia/gateway/flowroute/<external 10digit number> &fifo('myque out wait') uuid_broadcast <uuid from previous command> <path_to_audio_file> I know that I can have an audio file playing within the fifo arguments but was desiring more interactive control that uuid_broadcast looked like it could provide. Thanks, Lynn -- -- Lynn Nielson Green Seed Technologies / The RedX

1 0

Emulate nat=yes
by Emrah 16 Aug '12

16 Aug '12

Hello, I am having trouble connecting my Polycom phone to FS. It works fine with Asterisk and I am trying to emulate the nat=yes behavior over on FreeSWITCH. Both the Asterisk and FS servers are public. I am using the default FreeSWITCH sample configuration and attempting to register on the internal profile. I have tried using NDLB-connectile-dysfunction with no luck. Any suggestions of what else I could try? Directions to CLI commands to debug SIP would be much appreciated as well. Thanks a bunch for your help. -- Emrah “In theory, theory and practice are the same. In practice, they are not.” Albert Einstein

1 0

Regarding outgoing call
by Arnuld Uttre (Phonologies) 16 Aug '12

16 Aug '12

Currently I am making an outgoing call using "api originate" over telnet login. I am subscribing to: 1) CHANNEL_CREATE to know when a call is started. 2) CHANNEL_DESTROY to know that call is over and all the resources are freed. is this the right method ? NOTE: someone told me I should use CHANNEL_ORIGINATE to know that a call is started and CHANNEL_HANGUP_COMPLETE to know when call is over and all resources are destroyed. Wiki does not have much information on this. That is why I posted this question. -- Arnuld Uttre Systems Software Engineer arnuld(a)Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992

1 0

IP authentification
by Jakub Tencl 15 Aug '12

15 Aug '12

Hi, i'm in the middle of the testing our fresh installation and now i can't find out howto set IP authentification, when freeswitch are registered on the GW, it works fine but without registration, just based on IP i have no idea how the freeswitch is doing that if so. Can you help me please? I am using the vBilling system. Thanks Jakub

2 1