January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Distinguishing Blind REFER from application transfer
by mayamatakeshi 24 Jan '10

24 Jan '10

Hello, I'm using mod_xml_curl to provide dialplan. In my application, I need to differentiate if a call has entered the dialplan again due to uuid_transfer or due to Blind Transfer. I know I can recognize a Blind Transfer by checking variable_sip_h_Referred-By and variable_sip_refer_to. However, these variables are not cleaned up when the call reenters the dialplan due to application transfer or uuid_transfer. I realize I can differentiate them by adding some prefix like this: uuid_transfer TRANSFER,DestinationNumber XML default Or it is my responsibility to call application set or uuid_setvar to unset the variable(s) in this case? it seems to me the bad thing here would be to have to add lots of preventive <action application="set" data="sip_refer_to=" /> in my dialplan. Before doing so I thought in ask if the REFER-related variables being preserved upon dialplan reentry would not be a bug (well, it could be a feature useful in some scenarios I suspect). br, takeshi

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Svar: Re: Svar: Re: Home setup with home company
by mailinglist 24 Jan '10

24 Jan '10

Ok, I set it up like this: But now it gives me: 2010-01-22 11:52:08.667564 [NOTICE] switch_channel.c:602 New Channel sofia/external/2680xxxx(a)87.54.25.116 [16baec2f-4407-df11-8fb3-000c29b7b4cb] 2010-01-22 11:52:08.800123 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->4692xxxx in context public 2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx(a)87.54.25.116 to XML[8203@default] 2010-01-22 11:52:08.830071 [INFO] mod_dialplan_xml.c:252 Processing 2680xxxx->8203 in context default 2010-01-22 11:52:08.830071 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/2680xxxx(a)87.54.25.116 to enum[8203@default] 2010-01-22 11:52:09.163439 [INFO] switch_core_state_machine.c:136 No Route, Aborting huh >>> Michael Collins 21-01-10 9:39 >>> On Wed, Jan 20, 2010 at 2:45 AM, mailinglist <mailinglist(a)fribert.dk> wrote: Hi Michael It's running on pfsense, so it's kinda locked to the version it currently is. Looks very nice though. Looking beyond that, is the action / anti-action list corrent? I would say that you need to add an anti-action under the day of week check and go to vm if it does not match. Right now if the DOW is 0 or 6 then the entire extension will "fail" and the dialplan will just move on. Remember that if any conditions fail then the entire thing extension "fails" unless you are doing interesting things with the break= parameter. See the dialplan page on the wiki for examples of how to use break in your conditions. -MC Best regards Fribse >>> Michael Collins 20-01-10 1:53 >>> On Tue, Jan 19, 2010 at 9:02 AM, mailinglist <mailinglist(a)fribert.dk> wrote: I have a very small one man constultancy company that has the occasional call, unfortunately we are getting more spam calls after hours than real calls during work hours, so I would like to set up a TOD system. First step for me is just playing with the TOD example, I've gotten this far: group/company@${domain_name}"/> My idea with this, was that in the time 9-17 mon-fri, it rings, the rest of the time it's sent directly to the voicemail. I would of course like to have it not take messages outside work hours, but that's just refining :-) But it picks up the call, and then nothing... We have a much cleaner way of doing TOD and DOW handling. You'll need to get to the latest FreeSWITCH version. Look in conf/dialplan/default.xml for this example: Use that condition instead of the two conditions you're now using and see if you have better success. -MC _______________________________________________ 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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Javascript self.session.getVariable
by Mouncif Benniane 24 Jan '10

24 Jan '10

I have the following in javascript: caller_id = self.session.getVariable("caller_id_number") for some reasons it returns: ReferenceError: self is not defined, I am following this page: http://wiki.freeswitch.org/wiki/Session_getVariable any ideas?

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Presence Not Working After Version 1.0.5pre9?
by Jerry Richards 24 Jan '10

24 Jan '10

Does anyone know why I do not see NOTIFY messages with presence status being sent out from FS for two Bria softphones? It used to work in my old version 1.0.5pre9. I upgraded Freeswitch on Jan 12 and Jan 20 and edited in my XML configurations, but I do not see the NOTIFY messages since version 1.0.5pre9. I have mostly default configuration and I added the manage-presence=true setting. I posted a pastebin at http://pastebin.freeswitch.org/11867. Can anyone tell why this isn't working? Best Regards, Jerry

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INCOMPATIBLE_DESTINATION?
by Sergey Okhapkin 24 Jan '10

24 Jan '10

I often see in FS log the following problem (bypass_media=true), SVN r16340: SDP sent out to gateway (INVITE): v=0 o=bandx-msw3 0 0 IN IP4 213.166.9.4 s=sip call c=IN IP4 213.166.9.6 t=0 0 m=audio 56032 RTP/AVP 0 8 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=silenceSupp:on - - - - SDP response from gateway (183 Session Progress): v=0 o=- 3473087019 3473087037 IN IP4 67.203.64.182 s=- c=IN IP4 67.203.64.182 t=0 0 m=audio 14116 RTP/AVP 0 a=sendrecv a=ptime:20 a=rtpmap:0 PCMU/8000 Freeswitch hangs up the call with hangup cause INCOMPATIBLE_DESTINATION. Where is incompatibility? There is common codec 0.

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Managing Presence on Gateways
by afshin afzali 24 Jan '10

24 Jan '10

Hi Guys, In the external profile (as in the internal) there is an option to enable presence functionality (with setting it to passive). My question is how does it mean presence functionality for a gateway which interfaces home domain to another one? Does it mean that the gateway could subscribe itself for some presence information in that domain in behaves of local users and relays them? Appreciate all comments, -- afshin

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Eavesdrop when using simring
by Wellie Chao 24 Jan '10

24 Jan '10

I have eavesdrop working fine on outbound calls and also inbound calls where there is a single DID per IP phone. When I have a DID that rings multiple extensions simultaneously, what is the best way to obtain information about which extension has picked up the call and store that using hash? I can set a variable before I issue the bridge action, like so: <action application="hash" data="insert/${domain_name}-spymap/646xxxyyyy-1000/${uuid}"/> <action application="bridge" data="user/1001@${domain_name},user/1002@${domain_name},user/1003@${domain_name},user/1004@${domain_name}"/> However, that doesn't tell me who actually picked up, so at best I can allow users to eavesdrop on the last incoming call to the main DID, not the last incoming call to a particular extension. Is there something I can do in the bridge that will cause it to set a variable once it knows which extension has picked up the call?

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Possibly Bug in mod_sofia
by afshin afzali 23 Jan '10

23 Jan '10

Hi, In sofia_reg.c line 190 : nua_handle_bind(gateway_ptr->nh, gateway_ptr->sofia_private); It seems the right handle be gateway_ptr->sub_nh. It causes every SUBSCRIBE message's response print "Gateway information missing" error. -- afshin

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Call for help on FSComm
by João Mesquita 23 Jan '10

23 Jan '10

FreeSWITCH™ers, I believe that everyone already heard about FSComm by now. Development is moving fast and features are being added very fast. Nonetheless, we need help to get this software the way we want. If you are a GUI Designer or have any design skills and want to contribute, please, get in touch with me by email, IM, IRC on #fscomm or even pigeon if you like. I believe everyone wants a sexy looking softphone with all the features FreeSWITCH™ is able to provide. Regards, João Mesquita FSComm Developer GTalk: jmesquita(a)gmail.com PayPal: jmesquita(a)gmail.com IRC: jmesquita on #fscomm @freenode

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CDR Gateways
by FERNANDO VILLARROEL 22 Jan '10

22 Jan '10

Dear All. I have defined various gateways in ~/sip-profiles/external My questions is if a gateway named foo that send calls from diferents IP address like x.x.x.x and x.x.x.y. How i can doing a group for both ip address where i can doing later a sql like accountcode=foo? In this moment if i need know the traffic of customer foo, i should doing two differents cdr, one for every ip address. It's fine or i can doing of different way? I hope anyone could me explain how i can doing. my gateway foo.xml <include> <gateway name="foo"> <param name="username" value="foo"/> <param name="password" value="foo"/> <param name="realm" value="x.x.x.x"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> foo1.xml <include> <gateway name="foo1"> <param name="username" value="foo1"/> <param name="password" value="foo1"/> <param name="realm" value="x.x.x.y"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> Both gateways foo and foo1 are the same customer my cdr_csv.conf.xml <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings>   <param name="default-template" value="sql"/>   <param name="rotate-on-hup" value="false"/>  <param name="legs" value="ab"/> </settings> <templates> <template name="sql">INSERT INTO v_cdr (v_id, caller_id_name, caller_id_number, destination_number, context, start_stamp, answer_stamp, end_stamp, duration, billsec, hangup_cause, uuid, bleg_uuid, accountcode, read_codec, write_codec, remote_media_ip, network_addr) VALUES ("{v_id}", "${caller_id_name}" ,"${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}" ,"${uuid}","${bleg_uuid}", "${accountcode}", "${read_codec}", "${write_codec}", "${remote_media_ip}", "${network_addr}");</template> The argument accountcode on my database is Blank or None for all records of gateways. Regards.

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