August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Configuration of freeswitch with Acme Packet Session Directors and BroadWorks
by Nathan Neulinger 27 Aug '13

27 Aug '13

Running into an issue with getting a configuration up and running with a provider. Setup is a broadworks backend, fronted by Acme packet directors at the provider. The trunk is set up with a pilot userid, and configured with the various TNs on the trunk. Inbound calls to each of the DIDs are working without issue. Outbound calls will only work at this point if they configure IP authentication as a workaround. Have tried with and without the extension-in-contact/extension parameters, which had no effect on the outbound calls, but broke inbound due to it mapping everything to the single extension. Net symptom is that on an outbound, it does an invite, gets a 100 trying, and then a 403 forbidden. It never gets a 407 proxy auth required in response to the invite. The register request/interaction works fine though. Does anyone have this type of setup deployed that can suggest any config options/etc? I don't have any details of the configuration on the provider side, but if there is something specific, I can ask. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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fifo recording, loopback and other evil stuff
by Wesley Akio 27 Aug '13

27 Aug '13

Hi all, As far as I can tell there is no way to use fifo_record_template with variables for on hook agents. I has been asked and asked again, with no easy answer. Keeping that in mind I went down a not so pleasant path: <member timeout="10" simo="1" lag="5">{fifo_member_wait=nowait,ignore_early_media=true,loopback_bowout=false,loopback/a1000/default</member> As you can see there is a loopback right there. The recording is done in the dialplan and things work as they should, to a point. Since the softphone of choice has more than one line and nobody seems to be able to disable it we use the limit app to avoid call waiting, hence the loopback_bowout=false, otherwise the limit is freed on bowout. Point being: Does someone know a way to record the outgoing call to a fifo on hook agent in a cleaner way? Best, Wesley Akio TuntsCorp.com

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is it possible to have FS forcilby unregiter a registered UA
by Vincent Xia 27 Aug '13

27 Aug '13

is it possible to have FS forcilby unregiter a registered UA, for example, user/1001? to go one step further, how can i do the unregister upon receiving another register request? i checked event list of the event socket but cannot find any register event, can anyone shed any light on this?

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FreeSwitch and Yealink
by Paul 26 Aug '13

26 Aug '13

Hi guys, Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up. short FS LOG: switch_ivr_bridge.c:475 Channel [sofia/internal/105(a)10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105(a)10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going. Internal extension to extension works fine (even if the extensions are at a different physical location and subnet). I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have). I have updated firwmare to their latest version (which in the comments say freeswitch ready) Wondering if anyone else had any experience with these, or has some thoughts? Thanks Paul

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Looking for a Digital Artist to help out with a small FreeSWITCH video project
by Ken Rice 26 Aug '13

26 Aug '13

Hey Guys, I¹m looking for someone with Digital Art and Video skills to help out with a little project to create a FreeSWITCH Video 15 to 20 second Title/lead in video... Anyone that can help on this or knows someone that might volunteer a little time to help with this please contact me off list Thanks! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Porting C AGI from Asterisk to Freeswitch
by Valter Nogueira 26 Aug '13

26 Aug '13

I have several agi's written in C which I would like to port to Freeswitch. What is the best approach? Buld it as a freeswitch module? Thanks, Valter

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Capture DTMF and play and detect speech in Lua
by Manuel Durán Aguete 26 Aug '13

26 Aug '13

Hello, I'm trying to capture DTMF when doing play_and_detect_speech: function capture_dtmf(session,input_type,data,arg) if input_type == "dtmf" then freeswitch.consoleLog("CRIT","Key pressed: " .. data["digit"] .. "\n") end end session:answer() session:sleep(1000) session:setInputCallback("capture_dtmf","true") session:execute("play_and_detect_speech", "phrase:wellcome detect:unimrcp {start-input-timers=false,no-input-timeout=5000,recognition-timeout=5000, speech-language=es-ES}builtin:grammar/digits") xml = session:getVariable('detect_speech_result') if xml ~= nil then freeswitch.consoleLog("CRIT", xml .."\n") else freeswitch.consoleLog("CRIT", "No result!\n") end The asr seems to work, but when no dtmf is captured in capture_dtmf. Is DTMF capture with input callback supported by play_and_detect_speech ? Need to use detect_speech ? Thank you. Regards.

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display the expected caller id and name
by Vincent Xia 26 Aug '13

26 Aug '13

while making call from FS console to user 1005 and 1006 with originate sofia/internal/1005%192.168.1.35 &bridge(user/1006) XML default 101 102 the displayed caller id and name at 1005 is 101 and 102, at 1006 it's Outbound Call and 1005%192.168.1.35(but %19 here is recognaized as an ASCII SYMBOL and the display looks like 1005?2.168.1.35 and it's not user-friendly), so im trying to maniuplate the caller id name and number, as originate {origination_caller_id_name=123}sofia/internal/1005%192.168.1.35 &bridge(user/1006) XML default 101 102 but this only affects the display at 1005, how can i have both 1005 and 1006 to display the expected caller id and name.

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FS Caller id behavior
by Grant Bagdasarian 26 Aug '13

26 Aug '13

Hello, Consider the following scenario: SIP Server --------> Freeswitch SBC --------> Carrier The SIP Server originates a call through FS SBC to the Carrier. FS bridges A and B Legs. The caller id is set by the SIP Server: 1) First attempt with caller id visible 2) Second attempt with caller id blocked (anonymous) I tested this, but in both attempts the caller id was shown on the called party's phone. The Remote-Party-ID of the SIP INVITE (B-Leg: FS to Carrier) in the second attempt was like this, because the A leg INVITE didn't contain a Remote-Party-ID header: Remote-Party-ID: "anonymous" <sip:31123456789@10.0.0.1>;party=calling;screen=yes;privacy=off. When I configure FS to always dial out anonymously by setting the below variables, the call is bridged anonymously. <action application="privacy" data="full"/> <action application="set" data="sip_h_Privacy=id"/> <action application="set" data="privacy=yes"/> In the case of dialing anonymously the Remote-Party-ID looks like this: Remote-Party-ID: "31123456789" <sip:31123456789@10.0.0.1>;party=calling;screen=yes;privacy=full. So I'm assuming the problem lies with the SIP Server which initially initiates the call. It doesn't set the Remote-Party-ID header (correctly). If it did set it correctly, and I didn't configure the variables for anonymous dialing in FS, would FS still dial out anonymously? Or will the Remote-Party-ID be reset by FS?

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Remove custom SIP headers
by Grant Bagdasarian 26 Aug '13

26 Aug '13

Hello, So I found this page which shows how to remove custom SIP headers by using the unset application and passing in the custom header name prefixed by sip_h_[HEADER]. Is there a configuration setting which can be set to disallow custom headers being copied when bridging calls? The scenario is as following; 1) Each call is going through a Kamailio SIP Proxy first, which adds some custom headers. 2) Kamailio routes the INVITE to one of our FS SBC's. 3) FS bridges the call to the carrier. When FS creates a new call leg towards the carrier, I would expect it to NOT copy the custom headers added by Kamailio, but it does. This is my dialplan: <extension name="internal_to_external"> <condition field="${sip_req_host}" expression="^10\.0\.0\.1$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="privacy" data="full"/> <action application="set" data="sip_h_Privacy=id"/> <action application="set" data="privacy=yes"/> <action application="bridge" data="sofia/external/${sip_req_uri}"/> </condition> </extension> Did I configure something wrong, or is this just normal behavior?

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