- FreeSWITCH-users - FreeSWITCH Mailing Lists

Giving up resync after 1024 bytes
by Aqs Younas 12 Dec '14

12 Dec '14

Hi, All I am newbie to freeswitch, while playing multiple stream with mod_shout my freeswitch logs flood with this type of errors. [layer3.c:454] error: big_values too large! [layer3.c:454] error: big_values too large! Note: Illegal Audio-MPEG-Header 0xbbb7df0d at offset 19830. Note: Trying to resync... Note: Hit end of (available) data during resync. 2014-12-12 12:35:42.564619 [WARNING] switch_core_file.c:230 File has 2 channels, muxing to 1 channel will occur. Note: Illegal Audio-MPEG-Header 0xbbb7df0d at offset 19830. Note: Trying to resync... Note: Skipped 1024 bytes in input. [parse.c:682] error: Giving up resync after 1024 bytes - your stream is not nice... (maybe increasing resync limit could help). Note: Illegal Audio-MPEG-Header 0xef0000a8 at offset 20857. Note: Trying to resync... Note: Skipped 64 bytes in input. Could someone please help me in understanding or solving this problem. Thanks

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TLS versions and PFS settings
by Daniel Ivanov 12 Dec '14

12 Dec '14

What are the options to enable PFS on TLS support, i found the ability added in a changelog from feb,2014, but can't find the corresponding config file params. Also can i specify a list of allowed ciphers?

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Controling playback recording position
by ik 12 Dec '14

12 Dec '14

Hello, How can I control the position of the recording that is played ? For example, go 10 seconds forward or backward, like with Asterisk's ControlPlayback cmd. Thanks, Ido

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Re: [Freeswitch-users] Incomplete UPDATEs after Attended Transfer
by Joseph Frazier 11 Dec '14

11 Dec '14

Hi everyone, I'm trying to do attended transfers on my FS 1.5.14 box and have the endpoints receive UPDATE requests indicating that the identity of the other leg has changed. I installed the v1.5.14 tag from source and uncommented the following line of sip_profiles/internal.xml: <param name="NDLB-force-rport" value="true"/> Other than the above line, my configuration is completely vanilla. I have three extensions registered: 1000, 1007, 1006. My scenario is: - 1000 calls 1007 - 1000 calls 1006 - 1000 attended transfers 1007 to 1006 Here is my fs_cli with the following debugging options: sofia global siptrace on > sofia loglevel all 9 > sofia tracelevel alert > console loglevel debug > fsctl debug_level 10 https://gist.github.com/joseph-onsip/de90a66afdf37994c334 After receiving the REFER, FS sends an UPDATE to the refer target, but not to the referee. Is there a configuration option I can set in order to have an UPDATE sent to the referee as well? Thanks, Joseph

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Upgrade 1.4.14 TLS issue
by Arsen 11 Dec '14

11 Dec '14

Hi guys, I've upgraded my FS from 1.2 to 1.4.14 without changes in the config files and TLS stopped working. I can't register any more (503 service unavailable error). I don't see any logs in the console, but with ngrep I can see that the traffic hits the server on port 5061. tls_version = sslv23 if I change to tlsv1 I can register but outgoing calls don't work Any ideas? -- Regards, Arsen.

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How to play custom ringback using outbound ES.
by Post 11 Dec '14

11 Dec '14

Hi everyone, This is probably very simple, but I am stuck. I am using simple perl script to listen for incoming connection from freeswitch. I am willing to learn how to send a custom ringback to legA user. So far I managed to play some wav file in the following way: $con->execute("pre_answer", "", $uuid); $con->execute("playback", "somefile.wav", $uuid); So this works ok. Now I am trying to send a custom ringback (here are my guesses): 1) $con->execute("set", "ringback=\$\${ru-ring}", $uuid); $con->execute("pre_answer", "", $uuid); doesn't work (hear silence on legA user, see 183 with SDP in fs_cli sofia trace) 2) $con->execute("set", "ringback=\$\${ru-ring}", $uuid); $con->execute("ring_ready", "", $uuid); doesn't work either (hear RBT, pre-programmed in endpoint, see 180 w/o SDP in fs_cli sofia trace). Please give some advice. Regards, Pavel.

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Immediate disconnect while setting up incoming
by Sudhanshu 11 Dec '14

11 Dec '14

The setup is a GSM gateway, where I am trying to bridge all incoming calls to the included sims to an external number. The logs are: https://pastebin.freeswitch.org/23714 <https://www.google.com/url?q=https%3A%2F%2Fpastebin.freeswitch.org%2F23714&…> While trying a call to one of the sims, a ring or two can be heard followed by an immediate busy tone, signifying disconnect, which is corroborated by the logs. Why is this sudden disconnect originating? Can someone kindly help? -- Sudhanshu

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Re - FreeSWITCH Lua CUSTOM events interrupting session:streamFile playback - is this possible?
by Andrew Keil 11 Dec '14

11 Dec '14

To FreeSWITCH users, After progressing with Lua inside FreeSWITCH I have a question that I cannot source an answer for within the current FreeSWITCH documentation. I have a requirement for an external application to process an event (via the event socket layer) then return the results back to the application while an audio file is played back in a loop to the caller. >From the current documentation this method allows for the event to be sent then the return event is "consumed" using a polling approach. However the caller is listening to silence. function poll() -- create event and listener local event = freeswitch.Event("CUSTOM", "ping::running?") local con = freeswitch.EventConsumer("CUSTOM", "ping::running!") -- add text ad libitum event:addHeader("hi", "there") -- fire event event:fire() -- and wait for reply but not very long local retevent = con:pop(1, 5000) if retevent then print("reply received") freeswitch.consoleLog("DEBUG", string.format("reply received: %s\n",retevent:getHeader("Result"))) return true end print("no reply") freeswitch.consoleLog("DEBUG", "no reply\n") return false end Some questions: 1) Is there a way to playback audio (eg. session:streamFile(...)) while this takes place, since currently session:streamFile(...) seems to be a blocking function (ie. Finishes when the audio file is played back completely)? 2) Is there a way to use session:setInputCallback(...) to handle an external CUSTOM event being returned (since this would then be able to interrupt the session:streamFile(...) just like it does for DTMF or speech recognition)? If you have a preferred approach to solving this then I am open to your suggestions. I appreciate any assistance that you can give. Kind Regards, Andrew Keil

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Mod perl using $session->setVariable inline
by Liam Farr 11 Dec '14

11 Dec '14

Hi, I need to set a variable using a perl script inline. Normally I would use $session->setVariable('some_var',$my_value); However I cant find any options to execute this inline / as an inline action. (I need to evaluate the variable later in the hunting phase of the dialplan hence need it to be an inline action). Is there a way to do this? Or is there another way to achieve the same result, e.g. hacking $session->execute("set","some_var=$my_value") to run as an inline action? Any help would be much appreciated, I've scoured the wiki, confluence and git repo and come up empty. -- Kind Regards Liam Farr

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Re: [Freeswitch-users] Twilio SIP trunk config - outbound call rejected
by Ahmed Habiba 10 Dec '14

10 Dec '14

Mainly in Twilio if you are using sip trunk you should always call with international code prefix like you should put +1 before number for USA. it looks like the number you are using in local format “ 5101112222 <sip:5101112222@mycompany.pstn.twilio.com>” it should "+xxx 510111 " From: Adelia C. <hexade(a)hotmail.com <mailto:hexade@hotmail.com>> To: "freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>" <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Date: December 10, 2014 at 7:45:40 PM GMT+3 Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Subject: [Freeswitch-users] Twilio SIP trunk config - outbound call rejected Any idea why this INVITE is rejected with 400 Invalid Phone Number by Twilio? The first INVITE asks for Proxy Authentication, that seems to pass and this is rejected. Thank you! INVITE sip:5101112222@mycompany.pstn.twilio.com <sip:5101112222@mycompany.pstn.twilio.com> SIP/2.0 Via: SIP/2.0/UDP 10.50.1.116:5080;rport;branch=z9hG4bKa8QvU1B6QmeKc Max-Forwards: 69 From: "unknown" <sip:4152221111@mycompany.pstn.twilio.com <sip:4152221111@mycompany.pstn.twilio.com>>;tag=6UetZr6rDZgHF To: <sip:5101112222@mycompany.pstn.twilio.com <sip:5101112222@mycompany.pstn.twilio.com>> Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1 CSeq: 68751485 INVITE Contact: <sip:gw+twiliotrunk@10.50.1.116:5080;transport=udp;gw=twiliotrunk <sip:gw+twiliotrunk@10.50.1.116:5080;transport=udp;gw=twiliotrunk>> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.5.12b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Proxy-Authorization: Digest username="4152221111", realm="sip.twilio.com <http://sip.twilio.com/>", nonce="f722ae99d0a818bc422baa5d8", cnonce="IA1gJvqvEg6mmXR6WsQ", opaque="e89d2328e07da020d75a", algorithm=MD5, uri="sip:5101112222@mycompany.pstn.twilio.com <sip:5101112222@mycompany.pstn.twilio.com>", response="e65d3dea5b1e3b35306cfa121ec2ae4d", qop=auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 268 X-FS-Support: update_display,send_info Remote-Party-ID: "unknown" <sip:1000@mycompany.pstn.twilio.com <sip:1000@mycompany.pstn.twilio.com>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1418153731 1418153732 IN IP4 10.50.1.116 s=FreeSWITCH c=IN IP4 10.50.1.116 t=0 0 m=audio 21366 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 -- SIP/2.0 400 Invalid phone number To: <sip:5101112222@ingenio.pstn.twilio.com <sip:5101112222@ingenio.pstn.twilio.com>>;tag=08602519_6772d868_5eef74bb-957c-486e-b261-2a2e63704d73 Via: SIP/2.0/UDP 10.50.1.116:5080;received=10.50.1.116;rport=5080;branch=z9hG4bKa8QvU1B6QmeKc CSeq: 68751485 INVITE Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1 From: "unknown" <sip:4152221111@ingenio.pstn.twilio.com <sip:4152221111@ingenio.pstn.twilio.com>>;tag=6UetZr6rDZgHF Contact: <sip:171.118.14.130:5060 <sip:171.118.14.130:5060>> Content-Length: 0 --

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