January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] switch_core_sqldb.c:840 SQL ERR
by Bob Smith 11 Jan '12

11 Jan '12

Awsome. Thanks ! > fixed in tree

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switch_core_sqldb.c:840 SQL ERR
by Bob Smith 11 Jan '12

11 Jan '12

Hi, I think it may be related to me updating to the latest git point (cfa926d7 ....add sub_host to the sip_registrations table to keep track of the original host). Haven't seen these messages before today, and don't really want to break anything more than it might already be, so could someone point me in the right direction as to what needs to be done ? 2012-01-11 12:30:59.466486 [ERR] switch_core_sqldb.c:840 SQL ERR: [select sip_registrations.sip_user, sip_registrations.sub_host, sip_registrations.status, sip_registrations.rpid, '', sip_dialogs.uuid, sip_dialogs.state, sip_dialogs.direction, sip_dialogs.sip_to_user, sip_dialogs.sip_to_host, sip_presence.status,sip_presence.rpid,sip_dialogs.presence_id, sip_presence.open_closed,'','' from sip_registrations left join sip_dialogs on sip_dialogs.presence_id = sip_registrations.sip_user || '@' || sip_registrations.sub_host or (sip_dialogs.sip_from_user = sip_registrations.sip_user and sip_dialogs.sip_from_host = sip_registrations.sub_host) left join sip_presence on (sip_registrations.sip_user=sip_presence.sip_user and sip_registrations.orig_server_host=sip_presence.sip_server and sip_registrations.profile_name=sip_presence.profile_name) where sip_dialogs.presence_id='1000(a)an.example.com' or (sip_registrations.sip_user='1000' and (sip_registrations.orig_server_host='an.example.com' or sip_registrations.sub_host='an.example.com' or sip_registrations.presence_hosts like '%an.example.com%'))] no such column: sip_presence.sip_server Thanks Bob

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Provider Configuration: Vono
by Leonardo P. Bidinoto 11 Jan '12

11 Jan '12

Hi All. I was having a problem configuring a vono provider into my FreeSWITCH. Now its working. If someone had the same problem as me, here it's configuration. In external.xml file, insert the below line into settings: <param name="user-agent-string" value="FreeSWITCH Agent"/> Somehow, Vono Provider wont accept the default string that FreeSWITCH send. ex: FreeSWITCH-mod_sofia/1.0.head-git-db5f504 2011-07-17 17-00-38 -0400. and then, the vono.xml file: <include> <gateway name="vono"> <param name="username" value="[USERNAME]"/> <param name="realm" value="vono.net.br"/> <param name="password" value="[PASSWORD]"/> <param name="caller-id-in-from" value="false"/> <param name="extension" value="[USERNAME]"/> <param name="extension-in-contact" value="true"/> <param name="proxy" value="vono.net.br"/> <param name="register-proxy" value="201.86.87.2"/> <param name="expire-seconds" value="3600"/> <param name="register" value="true"/> </gateway> </include> Hope it helps someone. -- Leonardo Pires Bidinoto Voice Technology www.voicetechnology.com.br

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valet park hanging up on 1 side
by Rodney 11 Jan '12

11 Jan '12

i am trying to allow a "phone roulette" situation where two callers from the ivr can be randomly be joined together. I need the following options 2 for next caller 0 go back to ivr menu the matchup seems to work but once in a bridge the caller who press 2 moves on but the other caller gets hung up on from the whole system and has to call back. is there a better way to match callers in this scenario without the hangup. here is the extension: <extension name="valet_parking_destination_ext" > <condition field="destination_number" expression="^533$"> <action application="answer"/> <action application="clear_digit_action" data="all"/> <action application="bind_digit_action" data="moderator,0,exec:transfer,399 XML default"/> <action application="bind_digit_action" data="moderator,2,exec:transfer,533 XML default"/> <action application="playback" data="C:/Program Files/FreeSWITCH/recordings/randomchatsimple.wav"/> <action application="valet_park" data="valet_lot 533"/> <action application="transfer" data="533 xml default"/> </condition> </extension>

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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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