May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Setting up callbacks
by Alex Lake 07 Jun '13

07 Jun '13

I was wondering where's a good place to find some examples of how, by sending the right commands to the event_socket, I could have Freeswitch establish callbacks for me. Essentially there are a couple of different termination types - PSTN (via a gateway) and internally registered SIP accounts I would wish to be able to set up A->B (and maybe small conferences) using these types of destination in any combination. I've looked through the event_socket pages and the "originate" syntax, but would like to put together something a little more "idiot-friendly" so am looking around for precedents/tips...

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Is it possbile to remove Allow, Supported, Allow-Events from header in freeswitch
by divyeshkamothi 05 Jun '13

05 Jun '13

hi all there is requirement where i need to remove Allow, Supported, Allow-Events. from fs SIP headers. please provide me way using which i can remove. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Is-it-possbile-to-remove-Allo… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Can't change external RTP IP
by Oleg Stolyar 04 Jun '13

04 Jun '13

Hi guys, I am trying to change expternal_rtp_ip for one of my sip profiles. To make thing as simple as possible, I changed it in the internal profile (internal.xml). I just set it to a literal IP address. Then I registered a soft phone on the FS and tried to make a call. The call went through and FS console said that it connected to vociemail but I could not hear anything. Wireshark said that the SDP address that FS sent back is still FS's internal IP address. Any ideas what we are doing wrong? Thank you *Oleg*

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Encrypted RFC2833 DTMF
by afshin afzali 03 Jun '13

03 Jun '13

Hi Guys, I don't know if FreeSWITCH is capable of to exchange encrypted rfc2833 dtmf digits. My voip provider uses a Huawei switch and does ask me to enable dtmf encryption but I could not find such parameter in sofia.xml.conf file. Appreciate all comments Afshin

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Errer
by B.Tietz＠pinguin.ag 03 Jun '13

03 Jun '13

Hi, I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding UDPTL (X bytes)" with FS 1.2.10 with t38_gateway. Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 rxfax ... FS 1.2.5.3 works! Dialplan is <extension name="Incomming"> <condition field="network_addr" expression="^1.2.3.4"/> <condition field="destination_number" expression="^(01234.+)$"> <action application="set" data="num=$1"/> <action application="set" data="codec_string=PCMA,PCMU,G729"/> <action application="set" data="continue_on_fail=true"/> <action application="export" data="t38_passthru=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="bridge" data="{sip_execute_on_image='t38_gateway self nocng'}sofia/intern/${num}@192.168.1.1:41000"/> <action application="hangup" data="USER_BUSY"/> </condition> </extension> Anything I can try?! VG, Benjamin T.

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CHANNEL_HANGUP vs CHANNEL_HANGUP_COMPLETE
by Tihomir Culjaga 31 May '13

31 May '13

hello im wondering what should we track as hangup event ... CHANNEL_HANGUP or CHANNEL_HANGUP_COMPLETE ... i have a situation where A calls FS, an ESL application answers this call, originates a new call to B and bridges A and B... than ESL originate another call towards C and joins all 3 channels into a conference. now , A hangs up, FS sends just CHANNEL_HANGUP event ... not CHANNEL_HANGUP_COMPLETE ... im wondering if it is supposed to be like that or we need to get CHANNEL_HANGUP_COMPLETE ? T.

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sip_append_audio_sdp
by Richard Brady 31 May '13

31 May '13

Hi folks I am having some trouble with the sip_append_audio_sdp variable. If I set it like: <action application="bridge" data="{sip_append_audio_sdp=a=fmtp:18 annexb=no}sofia/external/sip:x@example.com"/> It seems to have absolutely no effect. I'm pretty sure I have seen it working in the past, so now I'm wondering what the prerequisites are for it to work (i.e. what am I doing wrong?!). I am using G729 in pass-through mode and have: inbound-late-negotiation [true] inbound-codec-negotiation [generous] The expected behaviour is for the value of this variable to be appended to the local SDP on the B-leg. The actual behaviour is that nothing is appended. Any thoughts? Richard

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Forward custom header
by Mino Haluz 31 May '13

31 May '13

Hi, I know freeswitch is b2bua, but if there is some way how could I forward custom header added earlier by proxy from one call leg to another, it would be perfect. What I'm trying to achieve in dialplan is: <action application="set" data="sip_h_X-Source-IP=${sip_h_X-Source-IP}"/> and bridging to gw. So this header could be transparently forwarded, but it does not add anything. Am I doing something really wrong ? Thanks, Mino

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Specifying MOH
by Sean Devoy 31 May '13

31 May '13

HI, I see in the wiki it says: <https://wiki.freeswitch.org/wiki/Variable_hold_music> hold_music Per-channel hold music. Supports all audio formats and audio streams. The hold_music variable can also be set globally at vars.xml. Usage: <action application="set" data="hold_music=/sounds/holdmusic.wav" /> For multi-tenant environment, if you want to have a separate MOH for the phone with hold button (like Polycom) that utilizes RE-INVITE with no media ip addr (0.0.0.0) for hold, you can override the hold-music values in the sip profile parameter similar to the following example: <action application="bridge_export" data="hold_music=$${sounds_dir}/music/company-a.mp3"/> I set this in my dial-plan like this: <anti-action application="bridge_export" data="hold_music=hold_music=/usr/local/freeswitch/sounds/RBW_On_Hold.mp"/> I had an error on the console that said "File access error." I chmod'ed the MP3 file. Now I get no error, but complete silence on hold. Any ideas? Thanks, Sean

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Hey Guys Today 3PM EST Friday FreeForAll
by Ken Rice 31 May '13

31 May '13

Hey Guys don¹t forget to join us at 3PM EST (that¹s like NOON PST) for the FridayFreeForAll For those of you Voice Only, sip:888@conference.freeswitch.org as usual For those that want to join with Video, sip:888@mcu.freeswitch.org For those that have the option, try the video Bridge lets see if we can find a breaking point with load! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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