December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dingaling In Component Mode "How To Call JingleTo SIP"
by Ali Jawad 11 Aug '14

11 Aug '14

Dear All This might be a duplicate post, I am sorry if it is. Normally I get direct feedback from the list. But this time I did not get any reply. So I was afraid it might not have been delivered. First I would like to state that dingaling is working in client mode with gtalk perfectly. I can call from gtalk to my sip extension "1000". I have setup component mode now I am using FS with Openfire, open does show the following: 1 fs.jabber2.mydomain.net LibDingaLing gateway However stats on the jabber server are Session Statistics: Packets Received/Sent: 1/1 So the component is REGISTERED, so far so good my problem now is how to initiate a call from XMPP to a SIP extension. Using client mode I would do the following: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@mydomain to the jabber client "Jabbin" does not popup an "add contact" on XTEN. Anything I missed here ? Thanks for any hints or suggestions The Freeswitch server logs are below: 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" to="fs" version="1.0"></stream:stream> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" from="fs.jabber2.MyDomain" id="69fcaa7b"></stream:stream> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <handshake>72dac833190326bb82dddedae558b2b794df5e1b</handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <handshake></handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <iq type="get" id="970-504" from="component.jabber2.MyDomain" to="fs.jabber2.MyDomain"> <query xmlns="http://jabber.org/protocol/disco#info"></query> </iq> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <iq from="fs" to="component.jabber2.MyDomain" id="970-504" type="result"> <query xmlns="http://jabber.org/protocol/disco#info"> <identity category="gateway" name="LibDingaLing"></identity> <feature var="http://jabber.org/protocol/disco"></feature> <feature var="http://jabber.org/protocol/disco#info"></feature> <feature var="jabber:iq:version"></feature> <feature var="vcard-temp"></feature> <feature var="http://www.google.com/xmpp/protocol/voice/v1"></feature> <feature var="jabber:iq:last"></feature> </query> </iq>

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Grandstream GXP 2000
by Jonas Gauffin 18 Dec '11

18 Dec '11

Hello I have problems with TCP and GXP2000 phones. They work fine using UDP, but no calls arrive if I switch to TCP. I've upgraded to the latest GS firmware today to see it that helped, but it didn't. Have anyone else had problems with TCP? Regards, Jonas

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xml_curl and gateways
by Peter P GMX 21 Feb '11

21 Feb '11

According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to set up gateways dynamically, but I do net get it to work: I always get : "Invalid profile" My assumptions for a right xml answer back to FS are as follows, but I think at least one of it is false: 1. ) I start with <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> 2.) I do not want to change settings, so I do NOT add a <settings> tag 3.) I want to enhance an existing profile, so I add a profile tag (<profile name='external'>) 4.) I assume that I can have an existing profile "external" in the xml conf files and can mix additional parts dynamically into the same profile 5.) Within the profile I want to add gateways, so I add a gateway tag (<gateways>) 6.) Then I add the gateways 7.) I add the closing tags Which one of the assumptions if wrong? The request I receive is as follows: {"key_name"=>"name", "action"=>"directory", "tag_name"=>"configuration", "key_value"=>"sofia.conf", "controller"=>"xml_curls", "hostname"=>"freeswitch", "section"=>"configuration"} I only answer once at startup with the follwing: <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> <profile name='external'> <gateways> <gateway name="freeswitch2"> <param name="username" value="1007"/> <param name="realm" value="192.168.0.56:5061"/> <param name="password" value="xxxx"/> <param name="proxy" value="192.168.0.56"/> <param name="expire-seconds" value="30"/> <param name="register" value="true"/> <param name="register-transport" value="tls"/> <param name="retry-seconds" value="30"/> </gateway> </gateways> </profile> </configuration> </section> </document> Does anybody have a correct XML answer for a gateway and can post it? Best regards Peter

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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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OSP Interop w/ Trans Nexus
by Gregory Boehnlein 02 Dec '09

02 Dec '09

Anyone know if Freeswitch can work w/ OSP and use something like Transnexus for CDR/Rating?

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freeswitch as a session border controller
by Robert McNaught 30 Mar '09

30 Mar '09

Hi, I have been reading the recent news articles about freeswitch. I am particularly interested in using it to replace a session border controller we have (kagoor voiceflow 1000). Has anyone actually used this in production as a session border controller for NAT traversal? Thanks Robert McNaught

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dynamic conference
by Carole O. 19 Mar '09

19 Mar '09

Hello, I have done a small change in my dialplan which works but since I am new with FreeSWITCH I was wondering if this solution goes with the philosophy of the software or if it is absurd and there is a solution more adapted . I try to reproduce the following functionality: "A and B are on a simple call and decide to add C and have a conference. Later on they decide also to invite D..." In the dialplan I have added (based on the default dialplan): <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> ................ <anti-action application="bind_meta_app" data="1 a o transfer:: transf_call_${caller_id_number}"/> <anti-action application="bind_meta_app" data="2 a s transfer::-both transf_call_${caller_id_number}"/> ....... </condition> </extension> <extension name="transf_call_to_conf"> <condition field="destination_number" expression="^transf_call_(.*)$"> <action application="answer"/> <action application="conference" data="conf$1@conf_dynamic"/> </condition> </extension> Here, if A calls B then A can bridge both legs into a conference named "confnumberofA" by dialing *2. Then, A can put this call on hold and call C. - If C answers and agrees then A can press *1 in order to bridge C into the same conference "confnumberofA". A will come back into the conference which is still on hold. - If C does not answer then A will still be able to come back into the conference it puts on hold. My main problem is the name of the conference. Since everybody should be able to convert a simple call into a conference, the conference's name has to be unique each time. I have chosen to make it dependent on the caller number which is not perfect because then he is the only one which can bridge the call and add member. However, I do not have any other idea, maybe I have missed another possibility. I would be glad to get any critics you have, it would help me to better understand the fundamental concepts. Thanks, Carole -- View this message in context: http://www.nabble.com/dynamic-conference-tp21053181p21053181.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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LDAP Integration
by Vinicius Kobashi 02 Feb '09

02 Feb '09

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Sending SIP calls via outbound proxy
by Erick Johnson 28 Jan '09

28 Jan '09

i forgot to give you the pastebin URL http://pastebin.freeswitch.org/6379 > > I'm running latest trunk - Revision: 10682 > > I've been doing an ngrep on my external freeswitch SIP port and FS > is not sending any SIP packets anywhere when I run the following command. > Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are > logged. > > originate > 'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' > &echo() > > Also, just to be clear, when I remove ";fs_path=..." from the command > above a call > is set up normally to erick at ejjohnson.org and the SIP packets are > logged > to console. > > Thanks guys. > > > What SVN rev are you running? Also, could you do a SIP trace? > > TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch > > Pastebin the output of that and we'll take it from there. > > -MC > > > > On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson > > <erick at junctionnetworks.com> wrote: > > > Both: > > > > > > originate sofia/external/'erick at > > ejjohnson.org;fs_path=proxybeta.foo.net' > > > &echo() > > > originate sofia/external/erick at > > ejjohnson.org;fs_path=proxybeta.foo.net > > > &echo() > > > > > > produce the exact same result & log > > > > > > :( > > > > > >> * I think you need to '' the sofia uri /b > > > > > > > > -- Erick Johnson sip/email erick(a)junctionnetworks.com 1-800-801-3381 x7006 Software Engineer Junction Networks

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