- FreeSWITCH-users - FreeSWITCH Mailing Lists

PBX
by Miha Zoubek 11 Jan '12

11 Jan '12

Hi, is it possible to have on freeswitch more than one PBX? If it is, where can I find this threat on wiki. Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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gsmopen - samsung gt-c3303k
by Nandy Dagondon 11 Jan '12

11 Jan '12

hi to gsmopen developers, fyi, i get this log message repeatedly with this mobile phone Samsung Champ GT-C3303K: AT+MMGL="HEADER ONLY" does not get OK from the phone. If your phone is not Motorola, please contact the gsmopen developers. Else, if your phone IS a Motorola, probably a long msg was incoming and ther first part was read and then deleted. The second part is now orphan. If you got this warning repeatedly, and you cannot correctly receive SMSs from this interface, please manually clean all messages (and the residual parts of them) from the cellphone/SIM. Continuing. tks, nandy

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Accept-contact in INVITE (RFC-3841)
by Giuseppe Grillo 11 Jan '12

11 Jan '12

Hi, What should I configure to include the Accept-contact parameter in INVITE message ? (ref. RFC-3841). Thanks, Giuseppe

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Re: [Freeswitch-users] NAT Confusion
by Bob Smith 11 Jan '12

11 Jan '12

Thanks Vitaly. Reckon I've got it sorted now.... >Make traces (tcpdump) at server and phone network, check all RTP streams >find where it get blocked (misdirected). most likely it is network related >problem.

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Multi-Tenant
by Miha Zoubek 11 Jan '12

11 Jan '12

Hi, I have done as is written on wiki, but my phones do not register. I guess that I have some problems with domains as I get this : 2012-01-11 11:31:15.430343 [WARNING] sofia_reg.c:2410 Can't find user [018108500(a)xxx.xxx.xxx.xxx] You must define a domain called 'xx.xxx.xxx.xxx' in your directory and add a user with the id="018108500" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2012-01-11 11:31:15.430343 [WARNING] sofia_reg.c:1359 SIP auth failure (REGISTER) on sofia profile 'internal' for [018108500(a)xx.xxx.xxx.xxxi] from ip xx.xxx.xxx.xxx xx.xxx.xxx.xxx is an Ip of my server. In my directory I have set domain kabelnet1. I have created kabelnet1.xml and dir kabelvoip1 in conf/directory. I have also commend in sip_profiles/internal.xml what is written on wiki. My question. If my domain for FS server in freeswitch.test.org and new created directory kabelnet1. Should the phone be registering and domain kabelnet1.freeswitch.test.org? Thanks! Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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User Configuration: Question about F_REC and misc.
by Dennis 11 Jan '12

11 Jan '12

hi, we would like to allow SIP-phones to connect to our fs directly. therefor we would like to be able to set some options for each user/phone. the most important setting is regarding the "record key" (F_REC), which one can find on some/most/all SIP-phones: 1.) is it possible to DISABLE the key for individual users? if we press the F_REC key on our snom, fs starts to record. we can find options to disable some keys, but we did not find an option to disable the F_REC key. this option is very important, because we can not allow everybody to record calls. 2.) addionally we would like to be able to play an audio-file, when the record key is pressed (to let the other side know, that the call is beeing recorded). misc. questions about user configuration options: can we set the DTMF-type (like info and/or rfc2833) or the allowed codec (like G722,PCMA@20i) to override the global settings? thanks a lot for your help! kind regards dennis

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voicemail_inject (Mod voicemail) problems
by Dennis 11 Jan '12

11 Jan '12

hi, we have problems getting "voicemail_inject" to work. whatever we do, we alway get the same error message (fs newest version). for example we do: voicemail_inject 1000(a)192.168.2.2 /usr/local/freeswitch/recordings/1000_2012-01-07-21-47-44.wav 1001 test we will always get: Error: [group=<group>[@domain]|domain=<domain>|<box>[@<domain>]] <sound_file> [<cid_num>] [<cid_name>] what are we doing wrong? thanks a lot for your help! kind regards dennis

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Dial by IP address
by Brian Foster 11 Jan '12

11 Jan '12

I have some phones on my network that are capable of dialing a SIP URI. It seems by default FreeSWITCH doesn't allow this, as it just picks up everything before the @ and puts it as the destination number. Is there a way to allow this? What would be needed in the dialplan? On Jan 10, 2012 9:05 PM, "John O'Brien" <john(a)whitesmiths.com> wrote: > Hi, > > We have a reasonably recent version of FreeSWITCH running. > FreeSWITCH version: 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) > We are running on CentOS 6 > > We start FreeSWITCH from a shell script. > > ulimit -s 240 > freeswitch -ncwait -u usrXXX -g grpXXX > > The issue is that things seem to go along nicely for a day to two. > Then without any apparent reason the freeswitch process dies. > Usually there is nothing happening when this occurs, system is idle. > > There appear to be no process remnants we can use for investigation. > > We would appreciate some suggestions on how we might track the issue down. > Is there some simple logging we can turn on etc? > > Regards, > > John O'Brien > john(a)whitesmiths.com > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 >

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Voicemail skip greeting
by Ryan How 11 Jan '12

11 Jan '12

Hi, I have a voicemail set up. I want it to not do the "Record your message at the tone, to end recording....". I just want it to go "My message... BEEEEEP". I've put skip_instructions=true in the dial plan, but it seems to have no effect. Anything I am doing wrong? <action application="set" data="skip_instructions=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> Thanks! Ryan

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Ota in sip
by bwibowo＠gmail.com 11 Jan '12

11 Jan '12

Dear all Is there any implementation of OTA in sip. Purpose is to upload new parameter when user connected to sip server. Thx Budi

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