- FreeSWITCH-users - FreeSWITCH Mailing Lists

how to pass session from dialplan xml to lua script?
by Vincent Xia 07 Jun '13

07 Jun '13

hi all, i have a lua script(record.lua) to record the call: record.lua -------------------------------------- callerid = argv[1] dest = argv[2] recording_dir = argv[3] filename = callerid ..dest recording_filename = string.format('%s%s', recording_dir, filename) if session:ready() then -- NOTE: the session here is nil! max_len_secs = 30 silence_threshold = 30 silence_secs = 5 test = session:recordFile(recording_filename, max_len_secs, silence_threshold, silence_secs); end but i don't know how to pass "session" from the default dialplan to lua in the following sentence: <action application="lua" data="record.lua ${caller_id_number} ${destination_number} $${recordings_dir}"/> any response is apprieciated!

4 3

Reinvite - Codec G.729 decoder error
by Kevin Mathy 07 Jun '13

07 Jun '13

Hi list, I have an issue in my Freeswich in SBC mode. It's a call from Freeswitch to my endpoint, my endpoint answer G729, after this 200OK, my end-point re-invite in G711A, and my freeswitch say: "Codec G.729 decoder error". Ex : 1) SIP Provider ----INVITE(PCMA, G729)-----> FS ----INVITE(PCMA, G729)-----> End-point 2) SIP Provider <-----(200 OK G729)------ FS <------(200OK G729)------ End-point 3) FS <------(Re-INVITE PCMA)------ End-point 4) FS Error "Codec G.729 decoder error" in logs In fact, it's something like http://jira.freeswitch.org/browse/FS-3739, but the patch doesn't seem to work in our situation. My freeswitch's conf: *sip_profile TOTO: * * * * <param name="dtmf-type" value="rfc2833"/>* * * * * * <param name="rfc2833-pt" value="101"/>* * * * <param name="inbound-late-negotiation" value="true"/>* * * * <param name="inbound-codec-prefs" value="PCMA,G729,PCMU"/>* * * * <param name="outbound-codec-prefs" value="PCMA,G729,PCMU"/>* *Diaplan TITI:* *<action application="export" data="codec_string=${ep_codec_string}"/>* *<action application="set" data="hangup_after_bridge=true"/>* * <action application="set" data="ignore_early_media=true"/>* *<action application="set" data="inherit_codec=true"/>* *<action application="export" data="t38_passthru=true"/>* *<action application="export" data="suppress_cng=true"/>* *<action application="bridge" data="* * {* * sip_invite_params=user=phone,* * sip_cid_type=none,* * ignore_display_updates=true,* * sip_renegotiate_codec_on_reinvite=true,* * }* * sofia/gateway/${distributor(LISTTMP ${sofia(profile TITIPRO gwlist down)})}/${numberdest}"/>* * * * * We choosed to use inbound-late-negotation with inherit-codec because of our t38_passthru option (as described here in FS wiki, I don't exactly remember where) To resolve this problem, we tried to find how to make FS forward the Re-INVITE received on B-Leg to the A-Leg, but we didn't found any solution... If someone have an idea, it would be great ! ;-) Thanks a lot, * Bien cordialement, Best Regards, **Kevin MATHY* * * --

1 0

What's the best language for implementing an IVR on Freeswitch
by Terry Ray 07 Jun '13

07 Jun '13

Greetings, While I've been very pleased with the stability and performance of FreeSWITCH itself, I'm having to scrap the Ruby implementation I was using to drive the call menu functionality for the IVR due to it's inability to maintain connectivity with FS via the inbound event socket. Would love some suggestions on the best approach. Lua? or some other approach? Mainly I'm looking for something that will be very stable at higher call volumes. Best regards, Terry

4 4

mod_voicemail + send_silence_when_idle
by Yuriy Nasida 06 Jun '13

06 Jun '13

Hi guys, One of my carriers drops a call if FS deos not send RTP media (10-20 sec). Thus issues happens, when voicemail module asks to leave a message. I tried to use send_silence_when_idle. I hear a noise from FS but not in the moment of recording... Please advice,Thank.

2 2

cannot make call, error INCOMPATIBLE_DESTINATION
by Khue Nguyen Minh 06 Jun '13

06 Jun '13

Hi All, I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me. Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200] Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200] Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200] Substituting codec PCMU@20i@8000h No audio codec available No 2833 in SDP. Disable 2833 dtmf and switch to INFO Send signal sofia/internal/963728839(a)x.x.x.x [BREAK] (sofia/internal/963728839(a)x.x.x.x) Callstate Change RINGING -> HANGUP Hangup sofia/internal/963728839(a)x.x.x.x [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Send signal sofia/internal/963728839(a)x.x.x.x [KILL] Send signal sofia/internal/963728839(a)x.x.x.x [BREAK] sofia/internal/963728839(a)x.x.x.x skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) (sofia/internal/963728839(a)x.x.x.) State EXECUTE going to sleep (sofia/internal/963728839(a)x.x.x.x) Running State Change CS_HANGUP (sofia/internal/963728839(a)x.x.x.x) State HANGUP Channel sofia/internal/963728839(a)x.x.x.x hanging up, cause: INCOMPATIBLE_DESTINATION Responding to INVITE with: 488 sofia/internal/963728839(a)x.x.x.x Standard HANGUP, cause: INCOMPATIBLE_DESTINATION Brs, Khue Nguyen.

4 7

Ring an extension on incoming calls
by Sherwood Hu 06 Jun '13

06 Jun '13

Hi all, We are considering to replace current PBX with freeswitch. I am new to this topic - and find that freeswitch is fantastic. Now I have a question - I setup the freeswitch to transfer to IVR extension when the main line number is called. Our current PBX first rings an extension (to allow the caller information displayed on the phone) then plays the IVR. I want this functionality in freeswitch too. How can I do so? Thanks in advance.

4 4

why disable_hold is not working?
by Vincent Xia 06 Jun '13

06 Jun '13

why disable_hold is not working? i set <action application="set" data="disable_hold=true"/> in the dialplan and also <param name="disable_hold" value="true"/> in sip profiles, nothing happened. another question is, disable_hold is described as a channel variable, i wonder how FS affects the channel by setting <action application="set" data="disable_hold=true"/> in the dialplan? for example, <extension name="disable_hold"> <condition field="destination_number" expression="^(11[01][0-9])$"> <action application="answer"/> <action application="set" data="disable_hold=true"/> </condition> </extension> in this extension, which one, the calling party or called party, will be disabled?

2 2

Freeswitch SDP manipulation with dynamic payload types
by Ashwin Rath 06 Jun '13

06 Jun '13

Hello all Here is our ingress SDP to freeSWITCH (partial text only) m=audio 40000 RTP/AVP 97 96 98 a=rtpmap:97 G.729b/8000 a=rtpmap:96 G.729ab/8000 a=rtpmap:98 G.729a/8000 and this is the egress from FS m=audio 20788 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Notice that FS is substituting the G729 flavors with just 18 in the outgoing SDP. Can we have the same codecs offered in the outgoing SDP as well ? --Ashwin

2 1

Multi network interface
by Khue Nguyen Minh 06 Jun '13

06 Jun '13

Hi all, I run FreeSWITCH on a server have two network interfaces. But, I see that, FS only listen/bind to one interface. I am using configure default of FS. How I can make FS listen/bind on two interfaces? Brs, Khue Nguyen.

3 2

How to get some RTP stats
by Mino Haluz 05 Jun '13

05 Jun '13

Hi, is there any way how to track: 1) packet loss 2) jitter 3) some other interesting RTP statistics in FS ? Thanks Mino

8 14