September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

simulate att_xfer , threeway
by Lee JJ 05 Sep '08

05 Sep '08

Hello : While the att_xfer , I collect show the calls and channels info . After press "0" , the calls is Zero , and found the last channel using application "three_way" # during att_xfer , press "0" freeswitch@TTN_Li> show channels API CALL [show(channels)] output: uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate 33c232f5-e0ff-446e-ad93-8c2f80d61656,2008-09-05 11:00:40,1220583640,sofia/inter2/1529@10.20.1.233,CS_EXECUTE,1529,1529,172.16.93.231,1526,att_xfer,sofia/inter2/1528,XML,inter2,PCMU,8000,PCMU,8000 0d51ddb6-cac2-428b-a310-d9c54f979efc,2008-09-05 11:00:40,1220583640,sofia/inter2/1526,CS_EXCHANGE_MEDIA,Extension 1529,1529,172.16.93.231,1526,playback,local_stream://moh,XML,inter2,PCMU,8000,PCMU,8000 501e6351-6b9f-46a6-81bc-9a2c5a3306fa,2008-09-05 11:00:53,1220583653,sofia/inter2/1528,CS_EXECUTE,Extension 1529,1529,172.16.93.231,1528,three_way,33c232f5-e0ff-446e-ad93-8c2f80d61656,XML,inter2,PCMU,8000,PCMU,8000 3 total. freeswitch@TTN_Li> freeswitch@TTN_Li> show calls API CALL [show(calls)] output: 0 total. freeswitch@TTN_Li> show calls API CALL [show(calls)] output: function,created,created_epoch,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid 2008-09-05 11:00:42,1220583642,switch_ivr_multi_threaded_bridge,1529,1529,1526,sofia/inter2/1529@10.20.1.233,33c232f5-e0ff-446e-ad93-8c2f80d61656,Extension 1529,1529,1526,sofia/inter2/1526,0d51ddb6-cac2-428b-a310-d9c54f979efc 2008-09-05 11:00:56,1220583656,switch_ivr_multi_threaded_bridge,1529,1529,1526,sofia/inter2/1529@10.20.1.233,33c232f5-e0ff-446e-ad93-8c2f80d61656,Extension 1529,1529,1528,sofia/inter2/1528,501e6351-6b9f-46a6-81bc-9a2c5a3306fa 2 total. freeswitch@TTN_Li> freeswitch@TTN_Li> show channels API CALL [show(channels)] output: uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate 33c232f5-e0ff-446e-ad93-8c2f80d61656,2008-09-05 11:00:40,1220583640,sofia/inter2/1529@10.20.1.233,CS_EXECUTE,1529,1529,172.16.93.231,1526,att_xfer,sofia/inter2/1528,XML,inter2,PCMU,8000,PCMU,8000 0d51ddb6-cac2-428b-a310-d9c54f979efc,2008-09-05 11:00:40,1220583640,sofia/inter2/1526,CS_EXCHANGE_MEDIA,Extension 1529,1529,172.16.93.231,1526,playback,local_stream://moh,XML,inter2,PCMU,8000,PCMU,8000 501e6351-6b9f-46a6-81bc-9a2c5a3306fa,2008-09-05 11:00:53,1220583653,sofia/inter2/1528,CS_EXCHANGE_MEDIA,Extension 1529,1529,172.16.93.231,1528,,,XML,inter2,PCMU,8000,PCMU,8000 3 total. So , what's the no calls changing meaning ? And I simulate the three_way working using scripts, pitty thing is either hanup immediately or someone in hold music. PLS give some advice . ### call test scripts staring n_sess = new Session() ; res = n_sess.originate(n_sess, "sofia/inter2/1528%10.20.1.233" ); // wait for it ... answer n_sess.waitForAnswer(10000); if ( n_sess.ready()) { //never go into here console_log("info", "n_sess uuid: "+ n_sess.uuid + "\n"); n_sess.execute("three_way", session.uuid ) ; }

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Re: [Freeswitch-users] Register cell phone (Gayatri Kulkarni)
by Steven Brown 05 Sep '08

05 Sep '08

I'm not sure about the Nokia 6085 but I have successfully used various mobile phones without a SIP client or wifi by using the Fring http://www.fring.com http://www.fring.com/downloado and GPRS/3G to connect to my FS box. If you have a good data plan this can be pretty economical. Hope this helps Steve Steven Brown steve.brown(a)justfone.com t. 08707706968 m. 07768755409 f. 07884636663 ---------------------------------------------------------------------- Message: 1 Date: Thu, 4 Sep 2008 22:17:06 +0530 From: "Gayatri Kulkarni" <xtpl.gayatri(a)gmail.com> Subject: Re: [Freeswitch-users] Register cell phone To: freeswitch-users(a)lists.freeswitch.org Message-ID: <76bef9440809040947y4207f75dj39dea8cf16ffc108(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Nokia 6085 On Thu, Sep 4, 2008 at 5:26 PM, hemant kumar <hemant.taan(a)gmail.com> wrote: > any information on the model and make of cell phone? > > On Thu, Sep 4, 2008 at 5:18 PM, Gayatri Kulkarni <xtpl.gayatri(a)gmail.com>wrote: > >> Hi, >> >> Can I register my cell phone to FS? >> How can I do that? >> >> Regards, >> Gayatri Kulkarni >> >> ----- >> Whenever you find yourself on the side of the majority, it is time to >> pause and reflect. >> >> _______________________________________________ >> 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-us >> ers >> http://www.freeswitch.org >> >> > > _______________________________________________ > 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-use > rs > http://www.freeswitch.org > >

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external call
by Gayatri Kulkarni 05 Sep '08

05 Sep '08

How do I call a number that is not administered on the FS, through FS? Regards, Gayatri Kulkarni ----- Whenever you find yourself on the side of the majority, it is time to pause and reflect.

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Gateway not registered
by Jair Santos 04 Sep '08

04 Sep '08

Hi, I have provider.xml on /usr/local/freeswitch/conf/directory/default <include> <user id="inphonex"> <gateways> <gateway name="inphonex"> <param name="username" value="3462101"/> <param name="realm" value="sip.varphonex.com"/> <param name="password" value="606434"/> <param name="proxy" value="sip.varphonex.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="3600"/> </gateway> </gateways> <params> <param name="password" value="606434"/> </params> </user> </include> but when I <mailto:freeswitch@maui> freeswitch@maui> sofia status gateway inphonex API CALL [sofia(status gateway inphonex)] output: Invalid Gateway! When I start FS I get this in the console 2008-09-04 09:50:04 [WARNING] switch_core.c:851 switch_load_network_lists() Cannot locate domain 192.168.1.117 Not sure if it could be related. Any help will be appreciated. thanks Jair Santos

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record_session sound quality
by Sune Kristensen 04 Sep '08

04 Sep '08

Hi I am having some problems with the quality of the recordings, but only for some calls. The FreeSWITCH server is running in Denmark and we have a phone provider in the UK (voipfone.co.uk). We have 2 external client connecting through OpenVPN, one in the UK and one in France. If I call the UK number and the UK client answers it, the quality of the call is fine, but the recording is very bad quality, almost impossible to hear what is being said. If that call is then transferred to the client in France (Or to one of the internal clients in Denmark), the quality is good again from that point forward in the same recording. I would think that the quality of the recording would be the same quality as the call itself, but that does not seem to be the case. Anyone has an idea about what can cause this and how to solve it? /Sune

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Conference pin behaviour
by Cavalera Claudio Luigi 04 Sep '08

04 Sep '08

Hello, I'm trying to understand this behaviour. First I create a new conference with this api command dialing towards myself: conference blah dial sofia/default/me@host I answer and it's ok, I'm the only user in the conference. then I set a pin for the conference API CALL [conference(blah pin 1#)] output: Pin for conference blah set: 1# then I invite a friend of mine to the conference with this api command: conference blah dial sofia/default/friend@host now my friend joins the conference without the need for a pin, is this correct? Maybe the pin are asked only to those joining the conference by themselves and not to those invited? Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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conference bind meta app
by Lee JJ 04 Sep '08

04 Sep '08

Hello : The bind meta no reponse , while conferencing . <extension name="nb_conferences"> <condition field="destination_number" expression="^(30\d{2})$"> <action application="answer"/>  <action application="bind_meta_app" data="4 b s javascript::calltest.js"/> <action application="conference" data="$1@default"/> </condition> </extension> Where is my bind meta syntax wrong ? thanks !

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CNG
by Tamas 04 Sep '08

04 Sep '08

Hello, I have this toplogy: CW--->FS--->SIPGW What I want to achieve is to send RTP continuously to CW as it cannot handle CN, but I would like to use CN between FS and that SIPGW (I don't know what is that equipment). Is this possible? I tried to play with supress_cng option but don't really know how should I manage this case. In sofia profile I have suppression commented out:  and I tried before bridge: <action application="set" data="supress_cng=true"/> and <action application="set" data="supress_cng=false"/> Any idea? Thanks in advance, Tamas

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tricky bridge
by Tamas 04 Sep '08

04 Sep '08

Hello, I have a sip pbx registered to FS. How can I send to that registered user call with defined called number? Example: I set userA with userApwd in directory. userA is registered to FS. I would like to dial number 1234 through that registration. How should the bridge data look like? Thanks in advance, Tamas

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registration information
by Sunil Singh 04 Sep '08

04 Sep '08

Hi, When I am starting my softphone I am not not able to see any kind of messages whether somebody is registring neither on conosole nor on 8015 port. Secondly what is the command to see registered users from console. regards, Sunil

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