- FreeSWITCH-users - FreeSWITCH Mailing Lists

svn to git
by Madovsky 20 Jul '10

20 Jul '10

Is there any special things to do to update last svn version to last git version ? Thanks Franck -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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ClueCon MMX - Time Is Running Out!
by Michael Collins 20 Jul '10

20 Jul '10

ClueCon is only two weeks away! This is a reminder that there are still a few rooms left at the Trump<http://cluecon.com/hotel-details>. When getting your room be sure to refer to booking number 4347 and ask for the discounted rate of $225 per night. This is a great value! ClueCon is the premier open source telephony developer conference in North America. Nowhere else can you enjoy such a great mix of technology and business professionals, a four-star hotel, and a price under $700. ClueCon has it all! Don't forget that we have a lot of great things<http://www.cluecon.com/node/48>to give away this year, including the laser-engraved <http://www.cluecon.com/node/47> iPads and Macbook Pro. It's not too late to sign up for ClueCon and get your hotel. To get registered or if you have questions, call 877-742-CLUE (2583). See you in Chicago! The ClueCon Team http://www.cluecon.com 877.742.CLUE (2583)

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Module for archiving all calls/conferences
by Farhan Husain 20 Jul '10

20 Jul '10

Hello guys, I want to record each call and conference that goes through FS and have the ability to later replay them. What is the best way to achieve this? Is there any existing module which can be used for this purpose with minor or no change? Should I write a module of my own from scratch? Thanks, Farhan

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Job Posting
by Joe Crivello 20 Jul '10

20 Jul '10

Hello all, I hope I'm sending this to the right list.. My company is currently looking to hire an experienced FreeSWITCH developer for a full-time position in Milwaukee, WI. The project you would be working on initially is green-field and pretty innovative. If you think you might be interested please take a look at the posting: http://www.milwaukeejobs.com/jobs.asp?pagemode=15&jid=2089191 Thanks! Joe Crivello

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Internal Registration Question (MWI?)
by Michael Gende 20 Jul '10

20 Jul '10

Hello All, I'm interested in understand more about how a user (1009@somedomain for instance) acquires that "somedomain". Here's an example: Registrations: ================================================================================================= Call-ID: b5429e1b9f1b91fd(a)192.168.99.101 User: 1009(a)*99.93.12.80* Contact: "user" Agent: Grandstream GXP2000 1.2.2.26 Status: Registered(UDP)(unknown) EXP(2010-07-20 11:54:30) Host: pfsense.local IP: 192.168.99.101 Port: 5060 Auth-User: 1009 Auth-Realm: *192.168.11.5* ================================================================================================= Note the differences in the User: domain (99.93.12.80 is actually the external IP address) and the Auth-Realm: 192.158.11.5 (which is the LAN ip of the FreeSwitch box). Typically, I'd like to see the user show as 1009@internalLANip or 1009(a)192.168.11.5 in this case. For some reason, I can't make that happen. Something stupid I'm doing, surely. But I can't see it at the moment. This functions fine for calling. But, I can't get the MWI to light up on the phone or get the tones to indicate a message. I assume this is because FS is looking for user@LANip not user@externalWANip. But the vm is there if I actually go look in the VM box. Comments? I'm scratching my head. Must be in the internal.xml somewhere. Regards, Mike

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Multiple CDRS for Bridged calls from an IVR
by David Allen 20 Jul '10

20 Jul '10

Hi, I have written an IVR that prompts a user to Enter a Number and then the call is bridged to a particular gateway. If the user wishes to end the call early I have set the bridge_terminate_key=# variable and continue_on_fail=NORMAL_TEMPORARY_FAILURE so the dial plan continues which loops back to the IVR. For collection of CDRs, I'm using XML-CDR. The problem I'm having is that each call bridged to the gateway is not producing an individual CDR record and is instead adding the CDR information as additional records under callflow in the XML when the call to the IVR is finally terminated, instead of when each bridged call is terminated. I need to get the CDR information as the call is terminated for credit control purposes or in otherwords when the bridge is termated for XML-CDR to post the calls CDR. I have looked on the Freeswitch site for a solution to this and I have not. Is there a way that I can get information when each call ends? Hopefully someone will be able to help with this question. Thanks David

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mod_cdr_csv changed?
by Apostolos Pantsiopoulos 20 Jul '10

20 Jul '10

Hi, I just updated my FS installation from a pre-1.0.6 one to the latest git snapshot and I am facing the following problem with mod_cdr_csv: When a call is successful I get only one cdr in Master.csv. That is the expected behavior since I have <param name="legs" value="b"/> in my conf. When the call is not successful I get 2 cdrs in my Master.csv. The dialplan is fairly simple : <context name="mytest"> <extension name="TEST"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="sofia/gateway/EXP2/777"/> </condition> </extension> </context> Is this expected behavior? In my pre-1.0.6 installation it didn't occur? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Re: [Freeswitch-users] How to change Header (From value) while originate
by Durmuş Ali Öztürk 20 Jul '10

20 Jul '10

Hello David, thank you for your answer. I have solved it by using the *sip_from_uri* parameter in the originate command like this. originate {sip_from_uri=sip:55555555@123.123.123.123<sip%3A55555555(a)123.123.123.123>}sofia/gateway/Test/44444444 &park() The caller Id is now changed to 55555555. Regards Ali ---------- Weitergeleitete Nachricht ---------- > From: David Ponzone <david.ponzone(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 19 Jul 2010 14:33:47 +0200 > Subject: Re: [Freeswitch-users] How to change Header (From value) while > originate > You should try to set sip_cid_type=none on the outbound leg. > And also, you will probably need this in the gateway config: > <param name="caller-id-in-from" value="true"/> > > David Ponzone Direction Technique > email: david.ponzone(a)ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pièces jointes sont confidentiels et établis à > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autorisée est interdite. Tout message électronique est susceptible > d'altération. **IPeva** décline toute responsabilité au titre de ce > message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas > destinataire de ce message, merci de le détruire immédiatement et d'avertir > l'expéditeur.* > * > * > > > > Le 19/07/2010 à 14:18, Durmuş Ali Öztürk a écrit : > > Hello, > > Using the *origination_caller_id_number *while doing a outbound call with > the originate command has no effect of the caller Id. > > Because, our VoIP provider demands the caller ID into the From tag (From: > <originator@domain>) of the SIP header, the *Remote-Party-ID* and * > P-Asserted-Identity* will be ignored. > > Is there an alternative way to manipulate the From field value (which is > taken from the username of a gateway profile by default)? Maybe another > parameter in a originate command? > > Thank you for your answer. > > Ali > _______________________________________________ > 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 > > > > > ---------- Weitergeleitete Nachricht ---------- > From: David Allen <david(a)virtutel.com.au> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 19 Jul 2010 23:18:19 +1000 > Subject: [Freeswitch-users] Multiple CDRS for Bridged calls from an IVR > Hi, > > I have written an IVR that prompts a user to Enter a Number and then the > call is bridged to a particular gateway. If the user wishes to end the call > early I have set the bridge_terminate_key=# variable and > continue_on_fail=NORMAL_TEMPORARY_FAILURE so the dial plan continues which > loops back to the IVR. For collection of CDRs, I'm using XML-CDR. > > The problem I'm having is that each call bridged to the gateway is not > producing an individual CDR record and is instead adding the CDR information > as additional records under callflow in the XML when the call to the IVR is > finally terminated, instead of when each bridged call is terminated. I need > to get the CDR information as the call is terminated for credit control > purposes or in otherwords when the bridge is termated for XML-CDR to post > the calls CDR. > > I have looked on the Freeswitch site for a solution to this and I have not. > Is there a way that I can get information when each call ends? Hopefully > someone will be able to help with this question. > > Thanks > David > > > > > > > ---------- Weitergeleitete Nachricht ---------- > From: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 19 Jul 2010 18:14:18 +0100 > Subject: Re: [Freeswitch-users] Multiple CDRS for Bridged calls from an IVR > Try setting log-b-leg to true in xml_cdr.conf.xml, that should then > generate CDRs for both the incoming call (aleg) and the outgoing calls from > the bridge (bleg) > > You can match them using the UUID, and identify whether a CDR is an a-leg > or b-leg from the value of <direction> (it will be either inbound or > outbound). > > Hope that helps. > > -Steve > > > On 19 July 2010 14:18, David Allen <david(a)virtutel.com.au> wrote: > >> Hi, >> >> I have written an IVR that prompts a user to Enter a Number and then the >> call is bridged to a particular gateway. If the user wishes to end the >> call early I have set the bridge_terminate_key=# variable and >> continue_on_fail=NORMAL_TEMPORARY_FAILURE so the dial plan continues >> which loops back to the IVR. For collection of CDRs, I'm using XML-CDR. >> >> The problem I'm having is that each call bridged to the gateway is not >> producing an individual CDR record and is instead adding the CDR >> information as additional records under callflow in the XML when the >> call to the IVR is finally terminated, instead of when each bridged call >> is terminated. I need to get the CDR information as the call is >> terminated for credit control purposes or in otherwords when the bridge >> is termated for XML-CDR to post the calls CDR. >> >> I have looked on the Freeswitch site for a solution to this and I have >> not. Is there a way that I can get information when each call ends? >> Hopefully someone will be able to help with this question. >> >> Thanks >> David >> >> >> >> _______________________________________________ >> 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 >> > > > _______________________________________________ > 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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FreeSWITCH and SS7
by Vitalii Colosov 20 Jul '10

20 Jul '10

Hi, Is it possible to interconnect FreeSWITCH with a PSTN operator using SS7 protocol? (they support only SS7 interconnect) If yes, what device could you recommend? Appreciate any hints! Thank you, Vitalie

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ESL
by Zuhair Raza 19 Jul '10

19 Jul '10

Dear list, I am trying perl esl by this link http://wiki.freeswitch.org/wiki/Perl_ESL my dialplan is <extension name="ivrd-example1"> <condition field="destination_number" expression="^99$"> <action application="log" data="INFO Let's do some ivrd, shall we?"/> <action application="set" data="ivr_path=/usr/local/freeswitch/conf/ ivr1.pl"/> <action application="socket" data="127.0.0.1:9090 full"/> </condition> </extension> when i call to the freeswitch box it reads the ivr file and hangs up the call and even doesn't print the line "Starting ivrd-hello_world.pl...\n\n" as written in ivr1.pl 2010-07-19 11:43:35.352121 [INFO] mod_dptools.c:984 Let's do some ivrd, shall we? EXECUTE sofia/external/5555(a)74.55.234.106set(ivr_path=/usr/local/freeswitch/scripts/ ivr1.pl) 2010-07-19 11:43:35.352121 [DEBUG] mod_dptools.c:854 sofia/external/ 5555(a)74.55.234.106 SET [ivr_path]=[/usr/local/freeswitch/scripts/ivr1.pl] EXECUTE sofia/external/5555(a)74.55.234.106 socket(127.0.0.1:9090 full) 2010-07-19 11:43:35.360110 [NOTICE] switch_core_state_machine.c:185 sofia/external/5555(a)74.55.234.106 has executed the last dialplan instruction, hanging up. 2010-07-19 11:43:35.360110 [DEBUG] switch_channel.c:2261 (sofia/external/ 5555(a)74.55.234.106) Callstate Change RINGING -> HANGUP my ivr1.pl is #!/usr/bin/perl # # ivrd-hello_world.pl # # Launched from FreeSWITCH using socket app and ${ivr_path} variable # use strict; use warnings; # Use lib is easy, but you can also install your ESL.pm and esl/perl/ESL/* into site_perl use lib '/root/freeswitch/libs/esl/perl'; use ESL::IVR $| = 1; # Turn off buffering select STDERR; # We'll take this stream, thanks print "Starting ivrd-hello_world.pl...\n\n"; ## Set the root dir for sound files my $sound_root = '/usr/local/freeswitch/sounds/en/us/callie'; ## Create the connection object which is basically an IVR my $con = new ESL::IVR; ## Get this channel's uuid and dialed number my $uuid = $con->{_uuid}; my $dest = $con->getVar('destination_number'}; ## Answer call, sleep for half second to let media get up and running $con->execute('answer','',$uuid); $con->execute('sleep','500'); ## Play a simple greeting $con->playback('/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav'); ## Tell the call what he/she dialed $con->execute('sleep','500'); $con->execute('say', "en number iterated $dest"); ## Politely disconnect $con->execute('sleep','1000'); $con->playback('/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-thank_you.wav'); $con->execute('sleep','500'); $con->playback('voicemail/vm-goodbye.wav'); -- Can any one suggest me from where i am going wrong.

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