- FreeSWITCH-users - FreeSWITCH Mailing Lists

[Semi-OT] MobiGater Pro in buy collective from Germany/France
by Michelle Konzack 28 Dec '10

28 Dec '10

Hello *, I am now in contact with Eurodesign which is the Manufacturer of "MobiGater Pro". Since hey have no disributor corrently in Germany and France and I am in creation of a new OnlineStore, I a thinking on sellig the MobiGater (all versions) in Germany and France. Since I am not very finace-stable currently I like to offer you a buy collective for a exceptionel special price. Minimum quantity is for me 12pcs/Karton and any Waranties are handled by "itsystems@tdnet UG i.G.". If you are interested please send a mail to <sales(a)itsystems.tamay-dogan.net> or me directly. Thanks, Greetings and nice Day/Evening Michelle Konzack -- ##################### Debian GNU/Linux Consultant ###################### Development of Intranet and Embedded Systems with Debian GNU/Linux itsystems@tdnet France EURL itsystems@tdnet UG (limited liability) Owner Michelle Konzack Owner Michelle Konzack Apt. 917 (homeoffice) 50, rue de Soultz Kinzigstraße 17 67100 Strasbourg/France 77694 Kehl/Germany Tel: +33-6-61925193 mobil Tel: +49-177-9351947 mobil Tel: +33-9-52705884 fix <http://www.itsystems.tamay-dogan.net/> <http://www.flexray4linux.org/> <http://www.debian.tamay-dogan.net/> <http://www.can4linux.org/> Jabber linux4michelle(a)jabber.ccc.de Linux-User #280138 with the Linux Counter, http://counter.li.org/

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Some outdated/obsolete modules?
by mazilo 28 Dec '10

28 Dec '10

While porting and compiling FS git (2010/12/27) to an ARM platform, I noticed the build/modules.conf.in has left out several modules, i.e. applications/mod_fax, applications/mod_limit, and codecs/mod_voipcodecs. After a little chat with stkn on the FS IRC channel, I learnt that these modules are no longer in used and have been considered obsolete. If so, why not start doing a housekeeping and remove the pertinent modules' source files from the circulation? This way, the obsolete modules won't cause any further confusion to new comers to FS git like it did to me. Just my 2¢. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Some-outdated-obsolete-module… Sent from the freeswitch-users mailing list archive at Nabble.com.

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call pickup
by samir 28 Dec '10

28 Dec '10

hello, I am trying call "pickup" for the incoming calls with intercept function but it do not seems to work, am i missing something ? Regards Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-pickup-tp5837233p5837233… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Time Conditions & Break - Weird Behavior? & Anti-actions Usefulness?
by Avi Marcus 27 Dec '10

27 Dec '10

I have two xml config files with time of day routing, and I couldn't imagine why when it reached the first one, it just skipped the rest of the xml files, as such: Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 parsing [default->YLE1] continue=false Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 Regex (*PASS*) [*YLE1*] destination_number(t1105) =~ /^t1105$/ break=on-false Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 Date/Time Match (*FAIL*) [*YLE1*] break=on-false 2010-12-27 01:36:09.410521 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/sip:1000@109.67.185.37:5072) State Change CS_ROUTING -> CS_EXECUTE But when I just edit the first and set continue=true, it suddenly now properly moves on. But it never matched the entire extension's condition tags, so why did it think it should not continue? (xml files here: http://pastebin.freeswitch.org/14885) Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 parsing [default->YLE1] continue=true Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 Regex (PASS) [YLE1] destination_number(t1105) =~ /^t1105$/ break=on-false Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 Date/Time Match (FAIL) [YLE1] break=on-false Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 parsing [default->YLE2] continue=false Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 Regex (PASS) [YLE2] destination_number(t1105) =~ /^t1105$/ break=on-false Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 Date/Time Match (PASS) [YLE2] break=on-false Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 Regex (FAIL) [YLE2] time-of-day() =~ /08:00-22:00/ break=on-false Dialplan: sofia/internal/sip:1000@109.67.185.37:5072 ANTI-Action voicemail(default ${domain} 1105) 2010-12-27 01:37:24.754671 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/sip:1000@109.67.185.37:5072) State Change CS_ROUTING -> CS_EXECUTE And - it seems the anti-action is triggered no matter how "off" we are, is there a way to have a less agressive anti action? E.g. it only activates if the first condition is true? It's stopping the rest of my dial plan from executing. When would one ever want an anti-action other than as his last <extension> in which case it could just be a new last extension with no condition tag. I don't understand why you would ever want them. Understanding would be appreciated :) -Avi Marcus

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Bad P-Asserted-Identity header from FS conference.
by M. Ranganathan 27 Dec '10

27 Dec '10

Hello, I am not sure what triggers this but I get a bad P-Asserted-Identity header from Free-SWITCH in an UPDATE request when I create a conference and send an INVITE to that conference. Here is the offending Request : UPDATE sip:502@192.168.5.75:6060;x-sipX-nonat SIP/2.0 Via: SIP/2.0/UDP 192.168.5.75:15060;rport;branch=z9hG4bK2ZtpZ94m4Zg7F Route: <sip:192.168.5.75:5060;lr;sipXecs-rs=*auth~.*from~OUQ1MENEMEQtODAzNjU5QkU%60.900_ntap*id~MTQ0MDMtOQ%60%60!34e982a809955a84b401940d3c785d61> Max-Forwards: 70 From: <sip:502@sipxtest.sipxtest.net;user=phone>;tag=2ecF2jHpjjH6c To: "user1" <sip:user1@sipxtest.sipxtest.net>;tag=9D50CD0D-803659BE Call-ID: 41745e29-669e33ca-619711cf(a)192.168.5.242 CSeq: 6399341 UPDATE Contact: <sip:502@192.168.5.75:15060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-2 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 297 P-Asserted-Identity: "conference" <conference> p-asserted-identity should specify a SIP URI and it does not. Is this indicative of a misconfiguration somewhere or is it a bug? Where is the P-A-I header constructed from? Thanks Regards, Ranga -- M. Ranganathan

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Time-of-day broken?
by Avi Marcus 27 Dec '10

27 Dec '10

This is my first day using time-based routing, and I'm not having fun. Regex (*FAIL*) [YLE3] time-of-day() =~ /00:00-07:59:59/ break=on-false freeswitch@internal> strftime 2010-12-27 02:14:41 This is on the latest GIT. -Avi

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mod_conference produces bad audio after commit 75198fe4cb130f3d6d0d43f50882188879a59f6f
by Peter Olsson 27 Dec '10

27 Dec '10

It seems like commit 75198fe4cb130f3d6d0d43f50882188879a59f6f causes bad audio within the conference. When commenting out some of the new code it works fine again. Jira is not up for the moment, but I will add a ticket there when it's up and running again. The changes below forces to use the timer (like it was before). -if (mux_used < bytes * 2) { +//if (mux_used < bytes * 2) { use_timer = 1; -} +//} /Peter Olsson

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GSM-Modems and support
by Michelle Konzack 27 Dec '10

27 Dec '10

Hello *, sometimes ago I already had asked but the answer was not very satisfait at the time so I ask now again: I want to build a local GSM-to-SIP-Gateway with 4 SIMs. Does FreeSwitch now support USB 1.1 GSM-Modems and if yes, which are recommened. I realy mean inexpensive non UMTS/HSPA ones. Thanks, Greetings and nice Day/Evening Michelle Konzack -- ##################### Debian GNU/Linux Consultant ###################### Development of Intranet and Embedded Systems with Debian GNU/Linux itsystems@tdnet France EURL itsystems@tdnet UG (limited liability) Owner Michelle Konzack Owner Michelle Konzack Apt. 917 (homeoffice) 50, rue de Soultz Kinzigstraße 17 67100 Strasbourg/France 77694 Kehl/Germany Tel: +33-6-61925193 mobil Tel: +49-177-9351947 mobil Tel: +33-9-52705884 fix <http://www.itsystems.tamay-dogan.net/> <http://www.flexray4linux.org/> <http://www.debian.tamay-dogan.net/> <http://www.can4linux.org/> Jabber linux4michelle(a)jabber.ccc.de ICQ #328449886 Linux-User #280138 with the Linux Counter, http://counter.li.org/

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how to know gateway used by distributor
by Octavio Duarte 27 Dec '10

27 Dec '10

Hello everybody I have this problem , when i use this <action application="set" data="continue_on_fail=true"/> 1 <action application="bridge" data="sofia/internal/${ distributor(gwusers)}%${domain_name}^$1" loop="4"/> or 2 <action application="bridge" data="sofia/gateway/${distributor(gwproviders)}/$1" loop="10"/> both work perfectly but i don't know for with gateway call went out i have tried with ${bridge_channel} the result for the first example is sofia/internal/sip:1002@1xx.142.1x7.xxx it is enough for me! but for the second example this is the result sofia/internal/5533186003 and i can't know the gateway that was used! i use this <param name="master-file-only" value="true"/> in cdr_csv.conf.xml so my question is: how can i know the gateway was used by distributor? thanks in advance for your help

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operator-extension in voicemail
by Madovsky 26 Dec '10

26 Dec '10

is operator-extension free to configure any extension transfer ? I'd like to us for example "9" while the voicemail is running to redirect in a conference extension Thanks Franck

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