December 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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FXO Card or Gateway
by Blake Priddy 20 Dec '13

20 Dec '13

Recommendation? Cost effective and reliable please :)

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help! did mapping to extension 1002 not working
by netandtech 20 Dec '13

20 Dec '13

I wand to achieve to incoming calls from did numbers to an extension So this is what I have with no success. In the default.xml <extension name="test_did"> <condition field="destination_number" expression="^(2281118003)$"> <action application="set" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="user/1002@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1002"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> In the public.xml <extension name="test_did"> <condition field="destination_number" expression="^(2281118003)$"> <action application="transfer" data="1002 XML default"/> </condition> </extension> And last I created a test.did.xml In freeswitch/conf/directory/default With the following <include> <user id="test_did"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="test_did"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="test_did"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension test_did"/> <variable name="effective_caller_id_number" value="2281118003"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> My log is the following 2013-12-20 12:12:56.376409 [NOTICE] switch_channel.c:968 New Channel sofia/internal/302281081619(a)109.68.164.21 [f80ee77e-6999-11e3-8083-11b99fb2bf0d] 2013-12-20 12:13:06.436421 [WARNING] switch_core_state_machine.c:514 f80ee77e-6999-11e3-8083-11b99fb2bf0d sofia/internal/302281081619(a)109.68.164.21 Abandoned 2013-12-20 12:13:06.436421 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/302281081619(a)109.68.164.21 [CS_NEW] [WRONG_CALL_STATE] 2013-12-20 12:13:06.436421 [NOTICE] switch_core_session.c:1506 Session 9 (sofia/internal/302281081619(a)109.68.164.21) Ended 2013-12-20 12:13:06.436421 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/302281081619(a)109.68.164.21 [CS_DESTROY] I was excepting a ring to my extension 1002 normally when someone calls the did number.. The did number should be mapped normally to the gateway account .. and not to my freeswitch server with extension 1002.. I am little confused here.. but I have left it normal setup for did number. Best regards

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SIP Trace Question / NAT one way audio problem.
by Sean P Devoy 20 Dec '13

20 Dec '13

Hi all, I seem to have fallen off the list! No mail since 7/26/2013. Have I made someone mad?? :) I have also been blissfully running with ISSUES what so ever since that time. I have a NATed client with PERIODIC one way audio. The client end has a Cisco 220W router with SIP ALG disabled. The phones are CISCO 504Gs and Polycom 330s(?), basciallt the same as everywhere else we have phones. I did a sofia global sip trace on for a while and captured the output. After writing an app to sort those sip packets into logical stream files, I dug in to the SIP conversations. This has left me with som questions for the people who know these things: 1. I found all my phones are REGISTERing every 23 seconds! That seems absurd TO ME, so where have I screwed that up/where do I set it and what is a reasonable number? 2. The only difference I see in the SIP packets from my working phone and these non-working ones is of course NAT related. The working phone's REGISTER looks like this: recv 757 bytes from udp/[WW.XX.YY.93]:5063 at 17:53:12.164203: ------------------------------------------------------------------------ REGISTER sip: <MYDOMAIN.COM> SIP/2.0 Via: SIP/2.0/UDP WW.XX.YY.93:5063;branch=z9hG4bK-238fcc8b;rport But the failing phones look like this: recv 801 bytes from udp/[AA.BB.CC.38]:5104 at 17:53:40.693691: ------------------------------------------------------------------------ REGISTER sip: <MYDOMAIN.COM> SIP/2.0 Via: SIP/2.0/UDP 10.2.2.239:5104;branch=z9hG4bKb374d6507894B431 The response is still sent to [AA.BB.CC.38]:5104 I guess first, IS THAT A PROBLEM? So, can anyone share any insight into the CISCO 220W config that I might be missing? Why would it only be a periodic problem? Could this be a problem where one end is using a wider range of rports than the other supports? Thanks, Sean --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

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