April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

hi
by virendra bhati 17 Apr '12

17 Apr '12

-- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati(a)gmail.com Skype id:- virbhati2 Hyderabad(India)

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transfering a call to another SIP server
by azza miled 17 Apr '12

17 Apr '12

Hi! I am using lua to make an IVR. I want to transfer the call to an another SIP server(OpenSIPS). Is this possibe? cause the wiki documentation just mention that we can transfer the call to a an another FS' extention. Thanks for the help.

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FreeSWITCH and Opensips
by azza miled 17 Apr '12

17 Apr '12

Hi! I want that my FeeSWITCH receives calls from my opensips' users without requiring authentication. In other words, I want that my FS receives unauthenticated calls, only from the IP address of my OpenSIPS server. any help, please???

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freetdm config question
by David George 17 Apr '12

17 Apr '12

Hello all, I just had a quick question on FreeTDM configuration. I have looked through the Wiki (both FreeSWITCH and Sangoma) and found a couple of references to running T1 boards with robbed bit signaling (RBS), but I have not gotten any configuration to work. Is anyone using FreeTDM with T1 RBS and able to comment on my freetdm.conf? [span zt tsu600e] fxo-channel => 1-4 I am using an Adtran channel bank (TSU600e) with FXS cards and a telephone to try to generate test calls (T1 configured as AMI/D4). This is connected to a Digium TE-220 Dual T1 card. I am using the Dahdi 2.6.0 drivers and I have no alarms on the T1. I am also using the latest FreeSwitch git (as of Apr 17 10am Eastern Daylight Time). Any help or pointers would be appreciated. Thanks, -- David George Hark Technologies

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dynamic dialplan alternatives
by huseyin kalyoncu 17 Apr '12

17 Apr '12

we are currently have two fs boxes load balanced by osips. in each fs we have dynamic dialplan with mod_xml_curl using php & apache & mysql. this conf is doing just fine with current load (about 20 cps) but with the increasing of incoming traffic, it shows some performance issues. we want to increase our call throughput. i searched through fs site and mailing list and came up with following options: 1) using lua or(python?) to serve dialplan instead of mod_xml_curl 2) writing a dialplan module(something like mod_xml_curl) in c which will do all the db lookups etc.. 3) using mod_erlang_event in outbound mode & spawning several erlang workers to do db lookups etc.. 4) using mod_event_socket in outbound mode & making db lookups on a different server. there is little information about 3 and 4 (only relative wiki pages) so im not sure even they are make any sense. before choosing one of above options i want to ask that which one of these would you prefer to increase performance. or do you have another suggestion other than these? thanks & best regards, Hüseyin Kalyoncu

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Voicemail hangups
by George Sapak 17 Apr '12

17 Apr '12

I have a quite a few people that want to leave voice mails and they say they are being hung upon as if they were not recording a message, I think its usually from people that are quitter speakers, these are my settings in my voicemail file: <param name="record-silence-threshold" value="1"/> <param name="record-silence-hits" value="999"/> is there a way to set the params so the vm records event if there is almost total silence Thank you, George.

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mod_com_g729 log file
by huseyin kalyoncu 17 Apr '12

17 Apr '12

hello, mod_com_g729 creates a log file named "1" under /usr/local/freeswitch which is containing lost of lines just like these: Success allocating G.729A/1 Success allocating G.729A/0 Success deallocating G.729A/1 Success deallocating G.729A/0 Success deallocating G.729A/1 is there any way to disable logging for mod_com_g729? regards, huseyin

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Pass the Regiseration message from SIP UA to remote SIP server
by fieldpeak 17 Apr '12

17 Apr '12

Could anyone advise how to configure FS to Pass the Regiseration message from SIP UA to remote SIP server like a SBC do, and once sip server down, the FS will keep the registeration list and take over the sip server job, thanks. -- Regards, Charles

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Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
by Daniel Knaggs 17 Apr '12

17 Apr '12

That's very interesting, Nathan. Now that makes three of us, two using the Digium cards and one using a Sangoma. It very much looks like something FreeSwitch is doing, perhaps the DTMF detection is far too sensitive? Are DTMF tones for an ISDN line inband or do they come in on the D-channel? Michael - any ideas what we can all do? From: Nathan Eisenberg [mailto:nathan@atlasnetworks.us] Sent: 13 April 2012 00:21 To: Daniel Knaggs Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls I'm having the exact issue you're describing, with the same card. However, what's interesting to me is that if I fire up Asterisk instead of freeswitch, the phantom DTMF problem disappears entirely. So it's *something* inside Freeswitch, at least for me. Would be interesting if you're able to do a similar test. Nathan Eisenberg Atlas Networks | Sr. Systems Administrator office: 206.577.3078 | www.atlasnetworks.us D Daniel Knaggs Software Developer Reality Solutions Ltd 1 Global Business Park Hamburg Road Kingston upon Hull East Yorkshire, HU7 0AE Tel: 01482 373104 Mobile: 07932 408313 Email: mailto:Daniel.Knaggs@realitysolutions.co.uk http://

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what is the purpose and value of the files in the db directory?
by Frank Church 17 Apr '12

17 Apr '12

I have been getting an occasional problem when freeswitch fails to start correctly. I googled around and found out that its was caused by the corruption of files in the db directory. What runtime related information is stored in those files? Are they useful for some logging or analytical value? Can an alternate database system be used for them? -- Frank Church ======================= http://devblog.brahmancreations.com

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