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Re: [Freeswitch-users] Add new headers with APIs
by Sayyed Mohammad Emami Razavi 23 Dec '13

23 Dec '13

Hello Alex, I used your guide in my lua script but in tcpdump and fs_cli, i can find no new headers set. This is my lua code: #!/usr/bin/lua require("ESL") local command = 'originate {ignore_early_media=true,originate_timeout=60}user/1000@192.168.1.5&playback(/usr/local/freeswitch/scripts/HelloWorld.wav)'; local args = table.concat({}, " "); local con = ESL.ESLconnection("localhost", "8021", "ClueCon"); con:execute("set", "sip_h_X-Token=abc123"); con:execute("set", "X-Token=abc123"); con:execute("set", "<![CDATA[sip_h_P-Charge-Info=<sip:${caller_id_number}@ ${domain_name}>;npi=0;noa=3]]>"); local e = con:bgapi(command, args); print(e:getBody()); Can you help me more?!

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About Freeswitch capabilities for video (calls, recording, etc...)
by Nick Techa 23 Dec '13

23 Dec '13

Hi, I'm trying to evaluate Freeswitch capabilities and determine if we can use it as an alternative to our current video calls platform (video calls, recording, live calls lawful tapping). The goal is support : 1)-audio/video calls recording support 2)-audio/video calls playback support 3)-audio/video calls lawful tapping (view/listen live calls) Video specific coverage in general seems very limited in books and docs. Assuming we'll use webrtc/sip and/or plain SIP, here is my question: Is Freeswitch currently able to do all the listed items? Any advice / suggestion or links for guides about how (or what might be needed) to support these items will be very helpul. Thanks.

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Dial plan public- HELP
by Andrew 22 Dec '13

22 Dec '13

Hi, I am building a standalone program where I will take an incoming call and do a lookup and then bridge the incoming call with an outbound call. I create a new session using the UUID that the in inbound call then I call the bridge code. What happens is as soon as I call the bridge call the inbound call disconnects. I know I need to edit the dial plan but not sure what to update it to. The other way I tried to do this is pass the bridge info via dial plan variable but the variable hits faster than I can set it. Is there a wait for variable I can put in the dial plan? Andrew

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Body on BYE message
by Sebastian Garcia 22 Dec '13

22 Dec '13

Hello, >From 1 year ago I'm FS user, at this moment I'm working in a system that send information in format XML to IP Phone screen via INFO SIP message through FS, periodically during a call established. To finish, I need to add a BODY on the BYE or 200 OK-BYE message, that contain the final information for the call in format XML too. I can add headers with the variable sip_bye_h_ but I can't add de final BODY to this message. Any suggestion are welcome. Thanks, Sebastian Garcia www.dattronic.com

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Add new headers with APIs
by Sayyed Mohammad Emami Razavi 22 Dec '13

22 Dec '13

How to add new headers to INVITE packet with APIs? e.g. my packet is this: INVITE sip:1000@192.168.2.73 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.77:5080;rport;branch=z9hG4bKUFpyDr890F1pB Max-Forwards: 70 From: "Outbound Call" <sip:TCAPI_User@192.168.2.73>;tag=UZ6gtcDZvQy8e To: <sip:1000@192.168.2.73> .... i want to add new attribute to these headers named 'Token', something like this: INVITE sip:1000@192.168.2.73 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.77:5080;rport;branch=z9hG4bKUFpyDr890F1pB Max-Forwards: 70 From: "Outbound Call" <sip:TCAPI_User@192.168.2.73>;tag=UZ6gtcDZvQy8e To: <sip:1000@192.168.2.73> Token: gasd^@@SGHT W454v$ERYWEVGR%^#V$CSDGASF .... .... How can i do this?!

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NAT problems, FS keep sending ACK to local IP Address
by Donny Hardyanto 21 Dec '13

21 Dec '13

Hi, I have “30 second-then-hangup” problem. I have traced the problem, and turn out that the server keeps sending ACK to SIP Client local IP Address (it is behind the NAT). Is it a misconfigure in FS? I notice that in SDP received from SIP Client the rtcp port is followed by SIP Client local IP address. Is it right or not? Here is the SIP trace result: recv 738 bytes from udp/[118.xxx.xxx.xxx]:6900 at 10:21:33.992037: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.150:57699 ;rport=57699;branch=z9hG4bK4Q9KyFrDr24gc From: "400001" <sip:400001@180.xxx.xxx.xxx>;tag=59etrFg3F0pvg To: <sip:400002@118.xxx.xxx.xxx:6900;line=b1bb925d6ab814e>;tag=189404928 Call-ID: ea145392-ddb9-1231-9c95-94de80208578 CSeq: 53085761 INVITE Contact: <sip:linphone.android@192.168.8.47:6900> Content-Type: application/sdp User-Agent: LinphoneAndroid/0.3.7 (eXosip2/3.6.0) Content-Length: 260 v=0 o=400002 1335 3400 IN IP4 118.xxx.xxx.xxx s=Talk c=IN IP4 118.xxx.xxx.xxx b=AS:380 t=0 0 m=audio 7078 RTP/AVP 98 101 a=rtpmap:98 speex/8000 a=fmtp:98 vbr=on a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7079 IN IP4 192.168.8.47 ------------------------------------------------------------------------ nta.c:3201 agent_recv_response() nta: received 200 OK for INVITE (53085761) nta.c:3264 agent_recv_response() nta: 200 OK is going to a transaction send 434 bytes to udp/[192.168.8.47]:6900 at 10:21:33.992228: ------------------------------------------------------------------------ ACK sip:linphone.android@192.168.8.47:6900 SIP/2.0 Via: SIP/2.0/UDP 180.xxx.xxx.xxx:57699;rport;branch=z9hG4bK502c0a9gNBU3Q Max-Forwards: 70 From: "400001" <sip:400001@180.xxx.xxx.xxx>;tag=59etrFg3F0pvg To: <sip:400002@118.xxx.xxx.xxx:6900;line=b1bb925d6ab814e>;tag=189404928 Call-ID: ea145392-ddb9-1231-9c95-94de80208578 CSeq: 53085761 ACK Contact: <sip:mod_sofia@180.xxx.xxx.xxx:57699> Content-Length: 0 ------------------------------------------------------------------------ nta.c:8169 outgoing_send() nta: resent ACK (53085761) to UDP/ 192.168.8.47:6900 Thanks, Regards Donny

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180 SDP to 183 (SDP)
by Mickael Hubert 21 Dec '13

21 Dec '13

Hi list, could you me explain, why Freeswitch "transforms" 180 (SDP) receive from leg B to 183 (SDP) to leg A ? Ex: INVITE --> FS --> INVITE .... 183 (SDB) <-- FS <-- 180(SDP) thanks in advance -- Cordialement HUBERT Mickaël Ingénieur VOIP - Hexanet --

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Outbound Audio Lost AFTER placing on Hold and picking back up.
by Sean Devoy 21 Dec '13

21 Dec '13

Hi All, I have a location that has been working solid for many months now, but the user reported a problem today. I can extract the FS LOG if anyone thinks there could be valuable data there, but unless requested I don't want to try to "redact" them. Also, this is a single occurrence problem so I am not ready to jump in the life rafts yet. But, this community frequently amazes me with solutions so it is worth asking. Here is what was reported to me: Yesterday at @ 3:38 had 2 calls come in. 1st call was fine until a 2nd call came in. First caller was put in hold; 2nd caller taken care of, then returned to 1st call. After 2nd caller disconnected 1st caller could hear Heidi but Heidi couldn't hear her then line disconnected. No. of that caller was 443-8xx-xxxx I read that as: 1. Call A answered. 2 way audio is fine. 2. Call A placed on HOLD. 3. Call B answered. 2 way audio is fine. 4. Call B hangs up. 5. Call A taken OFF HOLD. OUTBOUND AUDIO ONLY. 6. Call A disconnected???? (I am skeptical on this, how long do you sit on a NO AUDIO phone line before you hang up yourself.) The configuration is: FS on public IP <- ROUTER -> Internet <-NAT ROUTER -> Cisco SPA504G. I have no knowledge of the tech details of a line being placed on hold and then picked back up, so I hope you all have some ideas. Anything special for NAT concerns here? I do know that we have assigned each Line a separate port number of the SPA504G which helped with the initial setup. Anthony - I got your book BUT - My wife wrapped it for me for XMAS - grrrrr. Sean

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Gateways status on DB
by Chris B. Ware 20 Dec '13

20 Dec '13

Hi , I'm using odbc+mysql as "core DB". Freeswitch creates a table sip_registrations where I can see endpoints registration status. What about gateways? Should I use cli to get their status? Thanks Chris B. Ware

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Bad audio with playback application
by Ashwin Jain 20 Dec '13

20 Dec '13

Hi, We are facing issues with the bad audio quality (audio is sounding like its recorded underwater) only for the *playback application*. There is no issue with the audio call quality (after bridging). Audio file used with playback application is recorded at 8khz with 16 bits/sample and 1 channel. Following is the pseudo dialplan: 1. Call comes on freeswitch server. 2. Call answered. 3. Query database to identify destination 4. Play a welcome message audio (using playback application) 5. Start call recording 6. Bridge the call In step 4, we are facing this issue. -- Thanks and Regards, Ashwin Jain MetroGuild

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