- FreeSWITCH-users - FreeSWITCH Mailing Lists

How we can control calls using Java
by Adeel Ansari 13 Aug '08

13 Aug '08

Hi, I have managed to hook my Java program in. Now, looking for some hint, how we can control the call. Actually, I need to make a call, on genuine mobile phone, using Gizmo5 VoIP network. Furthermore, from where I can get the Mod-Java API documentation. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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Failing to execute OS Command
by Klaus Teller 12 Aug '08

12 Aug '08

Hi, I'm trying to convert recorded conversations from wav to mp3. I'm using ffmpeg for this purpose and i'm aiming to do this wthin Javascript. Yet, I'm encountering a problem. The conversion fails with the error message copied below. I must note that the same command works perfectly when executed from the command line. Any idea? Thanks, Klaus. ============ERROR MESSAGE============== 2008-08-12 15:29:03 [NOTICE] mod_dptools.c:1169 system_session_function() Failed to execute command: ffmpeg -i /usr/local/freeswitch/recordings/test.wav -ab 32kb /usr/local/freeswitch/recordings/test.mp3 2008-08-12 15:29:03 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 16 (sofia/external/4165465381(a)64.34.181.47) Ended 2008-08-12 15:29:03 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/external/4165465381(a)64.34.181.47 [CS_HANGUP] ==============ERROR MESSAGE========== -- Psssst! Schon das coole Video vom GMX MultiMessenger gesehen? Der Eine für Alle: http://www.gmx.net/de/go/messenger03

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Unable to receive calls from external sources.
by Rocky Hetherington 12 Aug '08

12 Aug '08

I'm having trouble getting FreeSWITCH to work properly and was hoping someone would be able to give me a few pointers. I've looked through all the documentation, examples and large chunks of the mailing list but i just can't make heads or tails of things. First i should give a bit of information on what i want to do. I'm going to be running FreeSWITCH on Xen virtual server with a public IPv4 IP. The OS is Debian Etch and lets call the machine "server1.example.org". It will be serving calls for the example.org domain and the relevant SRV records have been set up in the DNS. What i want to set up FreeSWITCH to do is have two users (we'll call them Bert and Fred). I want the two users to be able to call between themselves using either their relevant <USER>@example.org SIP URI or an extension number (2378 for Bert and 3733 for Fred). I also want users outside the domain to be able to call either user using their SIP URI and extension (and also via SipBroker). In addition Bert has two XMPP accounts. One is a Google Talk account and the other is a local account on an OpenFire server (also running on server1.example.org) I would like FreeSWITCH to listen on these accounts for incoming Jingle calls and then pass them through to Bert's registered SIP device. If the user is not currently registered i would like FreeSWITCH to take a voicemail. I would also like users to be able to dial out using SIP URIs and SipBroker prefixes. I don't currently need for it to dial out to the PSTN using a gateway. What i have done so far: Following the instructions in the Wiki i have created some Debian packages of FreeSWITCH 1.0.1. These are installed and the server is running fine. Other than the three changes outlined below the server is completely stock config. I have edited conf/vars.xml and changed the following line to: <X-PRE-PROCESS cmd="set" data="domain=example.org"/> I have deleted the local extensions from conf/directory/default and created two users called bert and fred using files such as this: conf/directory/default/bert.xml: <include> <user id="2378" mailbox="2378"> <params> <param name="password" value="password"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="accountcode" value="2378"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Bert"/> <variable name="effective_caller_id_number" value="2378"/> </variables> </user> </include> By default FreeSWITCH comes enabled to talk between internal users out of the box, so that works, although i need some way of aliasing the username "bert" to the extension 2378. I understand that FreeSWITCH uses something called contexts and the default context is not available to the public and i need to bridge to it, but i can't seem to figure out how to do it properly. I created a file conf/dialplan/extensions/bert.xml containing the following that i pieced together from the documention, but it doesn't seem to work: <include> <extension name="bert"> <condition field="destination_number" expression="^(bert|2378)$">   <action application="bridge" data="sofia/internal/2378%example.org" /> </condition> </extension> </include> (the commented out lines are previous lines i tried (one at a time) which also failed). I also tried adding the above (excluding the <include> tags) to the conf/dialplan/public.xml with no success. I then fire up my Gizmo softphone and attempt to connect from my Gizmo account using my SIP URI (bert(a)example.org) The FreeSWITCH logs indicate that it is hitting my server but is rejected: 2008-08-10 17:20:23 [DEBUG] mod_event_socket.c:1396 mod_event_socket_runtime() Socket up listening on 127.0.0.1:8021 2008-08-10 17:21:23 [DEBUG] sofia.c:3134 sofia_handle_sip_i_invite() IP 198.65.166.131 Rejected by acl domains. Falling back to Digest auth. 2008-08-10 17:21:23 [DEBUG] sofia.c:3134 sofia_handle_sip_i_invite() IP 198.65.166.131 Rejected by acl domains. Falling back to Digest auth. 2008-08-10 17:21:23 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [1747xxxxxxx(a)example.org] You must define a domain called 'example.org' in your directory and add a user with the id="1747xxxxxxx" attribute and you must configure your device to use the proper domain in it's authentication credentials. I'm unsure why FreeSWITCH believes that my Gizmo account is meant to be a local user, and am unable to find any documention on adding a domain to the directory. If anyone could just tell me how to allow external SIP calls to be passed through to local users i would be content to leave the voicemail, Jingle and outbound calls until a later time when i have a better understanding of FreeSWITCH. Hope someone can help. Thanks.

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SIP/2.0 407 Proxy Authentication Required
by Shehzad Pankhawala 12 Aug '08

12 Aug '08

Hi all, As H323 is not currently supported, I need to configure freeswitch to route calls to Yate2 to process my H323 calls and vice versa. For that I created the gateway in sip_profiles/external.xml <profile name="yate"> <gateways> <gateway name="192.168.1.23"> <param name="realm" value="192.168.1.23"/> <param name="extension" value="1001"/> </gateway> </gateways> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${default_codecs}"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="auto"/> <param name="sip-ip" value="auto"/> <param name="accept-blind-reg" value="true"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> </settings> </profile> _______________________________ Note xxx.23 is IP of yate and xxx.22 is ip of freeswitch I have also created the user in directory <include> <user id="4444" mailbox="4444"> <params> <param name="password" value="1234"/> <param name="vm-password" value="4444"/> </params> <variables> <variable name="accountcode" value="4444"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="4444"/> <variable name="effective_caller_id_number" value="4444"/> </variables> </user> </include> ______________________________________________ Now for the same freeswitch user I have created Yate Line on Yate server in accfile.conf (just like gateway in freeswitch) [MyFS] enabled=yes protocol=sip username=4444 description=FS account interval=600 formats=alaw,mulaw,gsm authname=4444 password=1234 number=4444 domain=192.168.1.22 registrar=192.168.1.22 To route calls from Yate to freeswitch I have created the folowing regexroute.conf entry of Yate server ^1001$=sip/sip:1001;line=MyFS; ;this means that calls dialed to 1001 will be routed to 1001 Freeswitch server. Then I dial from user registerd on Yate to 1001 The call is not processed of freeswitch at all, but while i put ngrep on port 5060 on server Freeswitch I found the following sequences of SIP messages: ___________________________________________ U 192.168.1.23:5060 -> 192.168.1.22:5060 INVITE sip:1001@192.168.1.22 SIP/2.0. Max-Forwards: 19. Via: SIP/2.0/UDP 192.168.1.23:5060;rport;branch=z9hG4bK1888370697. From: "4444" <sip:4444@192.168.1.23>;tag=735360950. To: <sip:1001@192.168.1.22>. Call-ID: 1932110763(a)192.168.1.23. CSeq: 25 INVITE. User-Agent: YATE/2.0.0. Contact: <sip:4444@192.168.1.23:5060>. Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO. Content-Type: application/sdp. Content-Length: 264. . v=0. o=yate 1218557236 1218557236 IN IP4 192.168.1.23. s=SIP Call. c=IN IP4 192.168.1.23. t=0 0. m=audio 19990 RTP/AVP 98 8 0 101. a=rtpmap:98 ILBC/8000. a=fmtp:98 mode=30. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=ptime:30. # U 192.168.1.22:5060 -> 192.168.1.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.23:5060;rport=5060;branch=z9hG4bK1888370697. From: "4444" <sip:4444@192.168.1.23>;tag=735360950. To: <sip:1001@192.168.1.22>. Call-ID: 1932110763(a)192.168.1.23. CSeq: 25 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.1-hacked. Content-Length: 0. . # U 192.168.1.22:5060 -> 192.168.1.23:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.23:5060;rport=5060;branch=z9hG4bK1888370697. From: "4444" <sip:4444@192.168.1.23>;tag=735360950. To: <sip:1001@192.168.1.22>;tag=4ayXjppg3X9Fe. Call-ID: 1932110763(a)192.168.1.23. CSeq: 25 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.1-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: 100rel, timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary. Proxy-Authenticate: Digest realm="192.168.1.22", nonce="ebe08c45-0762-435e-9986-28064ed91b10", algorithm=MD5, qop="auth". Content-Length: 0. . # U 192.168.1.23:5060 -> 192.168.1.22:5060 ACK sip:1001@192.168.1.22 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.23:5060;rport;branch=z9hG4bK1888370697. From: "4444" <sip:4444@192.168.1.23>;tag=735360950. To: <sip:1001@192.168.1.22>;tag=4ayXjppg3X9Fe. Call-ID: 1932110763(a)192.168.1.23. CSeq: 25 ACK. Max-Forwards: 19. Contact: <sip:4444@192.168.1.23:5060>. User-Agent: YATE/2.0.0. Content-Length: 0. . ________________________________________________________ Means calls are coming on freeswitch server, but even there is no process on freeswitch console. Why the calls are not processed on freeswitch, and displayed "SIP/2.0 407 Proxy Authentication Required." Is there any clue, mis-configured or any other solution, please reply. Thanks, Shehzad

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Re: [Freeswitch-users] [Sofia-sip-devel] SDP issue receiving calls from SIP connection
by Kirk Bateman 12 Aug '08

12 Aug '08

A partial solution, I've not figured everything out, but suspect it might cause issues elsewhere to do with codec selection, however ... I've added "." to the sdp_parse (parse_rtpmap)... so in the token call it allows ALPHA DIGIT "-." Works nicely now (so far), realised that all my codecs from the Tesco INVITE were just becoming G/<bitrate> Cheers Kirk 2008/8/12 Pekka Pessi <ppessi(a)gmail.com> > 2008/8/12 Kirk Bateman <kirk.bateman(a)gmail.com>: > > I've just noticed that all the rtpmap codec entries (from my Tesco > INVITE) > > generally have a "." in the token name ... it seems that sofia sdp_parse > > doesn't allow this. > > > > It seems valid according to the rfc 4566 ? > > The SDP parser has not been completely updated to RFC 4566. > > http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html#4566 > > -- > Pekka.Pessi mail at nokia.com >

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SDP issue receiving calls from SIP connection
by Kirk Bateman 12 Aug '08

12 Aug '08

Just played with the test_sdp tool ... looks like the issue might actually be the clock rate for format 125 ... tool says its invalid... Cheers Kirk

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how multiple sip profiles ?
by Lee JJ 12 Aug '08

12 Aug '08

Hello : Is there anyone tested multiple sip profiles ? >From the FAQ: >Q: How do I assign endpoints to contexts with different sets of extensions >Here are the different possible approaches: >Use 1 profile per context you want to route to (each one needs a distinct >ip:port) So I duplicate the sip profiles, internal , nat , external just give them another name and different ip to use . freeswitch@localhost> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= nat2 profile sip:mod_sofia@210.24y.y.y:5070 RUNNING (0) nat profile sip:mod_sofia@210.24x.x.x:5070 RUNNING (0) internal profile sip:mod_sofia@210.24x.x.x:5060 RUNNING (0) outbnd2 alias extern2 ALIASED outbound alias external ALIASED 210.24x.x.x alias inter2 ALIASED extern2 profile sip:mod_sofia@210.24y.y.y:5080 RUNNING (0) 210.24y.y.y alias inter2 ALIASED inter2 profile sip:mod_sofia@210.24y.y.y:5060 RUNNING (0) default alias internal ALIASED ^^^^^^^^ does this affect dialplan routing ? external profile sip:mod_sofia@210.243.126.69:5080 RUNNING (0) ================================================================================================= 6 profiles 5 aliases FYI: setting files sip profiles use the default config , and modify the correspoding parameter profile name , context , domain root@localhost sip_profiles# ls -F ./ ../ external/ external.xml inter2.xml internal/ internal.xml nat/ nat2.xml nat.xml  <profile name="inter2" domain="210.24x.x.x">  <aliases> <alias name="210.24x.x.x"/> <alias name="inter2"/> </aliases> root@localhost dialplan# ls -F ./ default.xml default.xml.old extensions/ inter2.xml ../ default.xml.simpl-test features.xml public.xml I also duplicate another pub2.xml for dialplan routing. freeswitch@localhost> sofia_contact 1526(a)210.24x.x.x API CALL sofia_contact(1526(a)210.24x.x.x) output: sofia/inter2/1526@59.124.94.250:2342 freeswitch@localhost> freeswitch@localhost> 2008-08-12 13:31:58 NOTICE switch_channel.c:533 switch_channel_set_name() New Channel sofia/inter2/1526(a)210.243.126.729b9ae75d-142d-4931-956f-c6624c2d61d6 2008-08-12 13:31:58 INFO mod_dialplan_xml.c:222 dialplan_hunt() Processing 1526->1527@default ^^^^^^ why always go to "default" context , not the "inter2" ? 2008-08-12 13:31:58 WARNING switch_ivr_originate.c:539 switch_ivr_originate() No origination URL specified! Please someone kindly advise !! Thanks

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Conferencing performance
by Fernando Testa 12 Aug '08

12 Aug '08

Hi list, I'm impressed by the quality of FS and I'm considering to suggest our company to use it for the conference platform. So, since I don't have currently a way to stress it, the concerns are: Q1: how is the quality under load? Q2 what are the limits of a conference bridge? Usually we get from 200 to 500 people in a system. could it be done? Q3 what about echo cancellation? Is that present on FS or is not an issue? (related to Q.1) Thank you! Fernando Gregianin Testa Voice Technnology Ltda

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DTMF recognition and nbound calls
by Ilan Perez 12 Aug '08

12 Aug '08

Dear All, I have setup my inbound extension (in the public.xml) diaplan to transfer to extension 5000 à the demo IVR The recordings come through nicely. From an internal call the demo fine including DTMF recognition but with inbound calls from external sourceie pstn line Two problems occur. 1. There is no DTMF recognition 2. If I hangup during the playback the system still plays out the whole IVR Here is my code for the extension. Hopefully someone can help me out <extension name="public_did"> <condition field="destination_number" expression="^(0283472006)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=C:/Program Files/FreeSWITCH/sounds/en/us/callie/voicemail/8000/vm-press.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2008 Diagnostic Devices, Inc."/> <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/> <action application="set" data="RECORD_ARTIST=Ian Curtis"/> <action application="set" data="RECORD_COMMENT=Love will tear us apart"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=false"/> <action application="set" data="playback_terminators=#*"/> <action application="record_session" data="C:/Program Files/FreeSWITCH/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_num ber}_${caller_id_number}.wav"/> <action application="set" data="ringback=${us-ring}"/> <action application="transfer" data="5000 XML default"/>  </condition> </extension> Anyone? Ilan Perez

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AppGen or GUI tool for FS ?
by Marek Gorecki 12 Aug '08

12 Aug '08

Hi FS community, I remember, that GUI issue, and actually many other FS control access issues are of high priority, and seems there is something what might/should help: http://adhearsion.com/ It is not straitforward since it was made with Asterisk in mind, but I believe it shouldn't be big issue to port it to FS too. I just trust in Anthony The Second persuasion and influence to convince Adhearsion authors/creators to make it also FS compatible - if not directly, then maybe they can open some API or other method of integration, and FS community directly, or via bounties, will bring it to reality. Its the first thought, before reading any further about Adhearsion. What are other opinions ? /\/\arekg

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