January 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

different mail-from/subject within voicemail
by Bernhard Suttner 06 Jan '11

06 Jan '11

Hi, is it somehow possible to specify the "email-from" address and subject used within the voicemail application within the directory (per user different)? Best regards, Bernhard

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old git
by Sam 06 Jan '11

06 Jan '11

Hi how can i download old git , if i want to download old git of git-34a0ca5 2010-12-24 20-38-57 -0600 Regards Sam

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Iksemel msvs compiling
by Norman Lam 06 Jan '11

06 Jan '11

Hi Everyone I am trying to compile the freeswitch with mod_dingaling and Iksemel to work on Google voice or Gmail voice, I have read This http://wiki.freeswitch.org/wiki/Iksemel_MSVS_project_example and the http://wiki.freeswitch.org/wiki/Dingaling But I am stuck at this clause Add HAVE_GNUTLS=1 the the preprocessor compile in the iksemel project What does this mean and how to go about this Thanks in advance Norman Lam

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Re: [Freeswitch-users] Can't build on Windows 7 64bit C++ Express!?
by Jeff Lenk 06 Jan '11

06 Jan '11

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Voicepulse destination_number
by Larry Wimble 06 Jan '11

06 Jan '11

Greetings Freeswitch gurus from a FS Newbie... I have a standard voicepulse account (4 channels) with 3 DIDs. In setting up Freeswitch for use with this account today, I was unable to make it route based on the dialed number. The relevant dialplan is this: <extension name="VP-IN-TEMP" > <condition field="context" expression="public"/> <condition field="destination_number" expression="ZZZZZZZZZZZ"> <action application="ivr" data="DEMO"/> </condition> </extension> When I set ZZZZZZZZZZZ to the number I'm dialing in on, it does NOT work. When I set ZZZZZZZZZZZ to my voicepulse login (from the credentials page), it works fine. It seems Voicepulse is transmitting my login ID where Freeswitch expects to see the dialed number. When I dial in, this appears in the log: 2011-01-05 19:51:50.672153 [INFO] mod_dialplan_xml.c:331 Processing Cell Phone FL <8135551212>->ZZZZZZZZZZZ in context public ...where ZZZZZZZZZZZ is my voicepulse login ID. The problem with this is that I can't route based on the number dialed. How do I find out what number was dialed? Any ideas? TIA, Larry Wimble

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Memory using
by Javier Galaz Jeria 05 Jan '11

05 Jan '11

Hello all I'm a new user of freeswitch and I've a few questions, 1. I've searched the mail list and some forums about the switch "waste" and I haven't found out what it does and how it does it. Anyone care to explain a little bit about it? The only thing that I've found is that it consumes more memory but in my tests [1] it doesn't change that much. 2. Do any of you guys have a rough estimate of what each call memory footprint should be? 3. About the ulimit -s 240, I've read that it limits the stack size, but I'm at a lost point as how to pick a suitable value. What is the criteria? 4. Are the tests that I've done [1] reasonable? I'm trying to know this things because I'm planning to use FreeSWITCH on a resource limited hardware. Best Regards Javier [1] test done with SIPp, using 0 to 20 users calling simultaneously, connecting the call, and hanging up after 5 secs. statistics using linear interpolation. using debian's init script (ulimit -s 240): slope 898.7 kB y-intercept 15410 kB using /usr/local/freeswitch/bin/freeswitch -waste -nc slope 810.6 kB y-intercept 16273 kB

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How to select audio from the initiator of a conference and play it to all participants
by Fabio Bigliardi 05 Jan '11

05 Jan '11

Hi all, I would like to configure a dialplan so that the initiator of a conference can select an audio file and play it simultaneously to all the participants in the conference. I tried the following: <extension name="my_mad_boss"> <condition field="destination_number" expression="^0916$"> <action application="set" data="conference_auto_outcall_caller_id_name=my mad boss"/> <action application="set" data="conference_auto_outcall_caller_id_number=0916"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(sales)}"/> <action application="bind_digit_action" data="cool,11,exec:playback,ivr/ivr-welcome_to_freeswitch.wav"/> <action application="bind_digit_action" data="cool,12,exec:playback,ivr/ivr-thank_you.wav"/> <action application="digit_action_set_realm" data="cool"/> <action application="conference" data="my_mad_boss@default+flags{endconf}"/> </condition> </extension> but it plays the right audio file only to the initiator of the conference. How to have it played to all members of the conference instead? Thank you in advance for your answer. F. Bigliardi

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FreeSWITCH Conference Call Today
by Michael Collins 05 Jan '11

05 Jan '11

Hey all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_05 We have a few things to discuss. Also, if DRK's schedule permits he will be dropping by to talk more about his billing stuff! Hope to talk to you soon. -Michael

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change nibble_account on answer
by Madovsky 05 Jan '11

05 Jan '11

..... <action application="set" data="nibble_account=000"/> ..... <action application="answer"/> <action application="set" data="nibble_account=111"/> ....... it seems that nibblebill keeps the nibble reference before answer ? Thanks

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SIP calls to Verizon FiOS?
by Avi Marcus 04 Jan '11

04 Jan '11

Hi, someone mentioned in IRC that they were finishing to submit paperwork for peering with Verizon. I have a few numbers on the FiOS network that I call often and would appreciate info on making SIP calls to them. Thanks! -Avi

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