July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to realize FS auth user (for REGISTERATION) using standard Radius interface
by fieldpeak 31 Jul '11

31 Jul '11

Dear frindes, i would to like to have all of users data in mysql db and autenticate user(for registeration) thru standard Radius interface e.g. FreeRadius, meaning when SIP user send REGISTRATION to FS, FS will send Radius request to FreeRadius, then FS auth user OK or NOK according to the response from FreeRadius. i read the wiki, there are some modules related, e.g. Mod_rad_auth, Mod_xml_radius, Mod_radius_cdr, however, no one provides a definite example for my case, could anyone help advise an example or any hint how to make it done, BTW, i can thought about using mod_xml_curl, however i don't want to use it if there is a better one. Thanks! Regards, Charles

1 0

Invitation to connect on LinkedIn
by srinivasulareddy kondreddy via LinkedIn 30 Jul '11

30 Jul '11

LinkedIn ------------ srinivasulareddy kondreddy requested to add you as a connection on LinkedIn: ------------------------------------------ Zohair, I'd like to add you to my professional network on LinkedIn. - srinivasulareddy Accept invitation from srinivasulareddy kondreddy http://www.linkedin.com/e/kwhdv8-gqqyne72-j/GrULditvq6UNXRBIaIGnrCzzv0aYgRN… View invitation from srinivasulareddy kondreddy http://www.linkedin.com/e/kwhdv8-gqqyne72-j/GrULditvq6UNXRBIaIGnrCzzv0aYgRN… ------------------------------------------ DID YOU KNOW you can showcase your professional knowledge on LinkedIn to receive job/consulting offers and enhance your professional reputation? Posting replies to questions on LinkedIn Answers puts you in front of the world's professional community. http://www.linkedin.com/e/kwhdv8-gqqyne72-j/abq/inv-24/ -- (c) 2011, LinkedIn Corporation

1 0

Load Balance Trunks
by Abid Saleem 29 Jul '11

29 Jul '11

Hi All, Please help me the following 2 issues. 1- I have 100 Trunks from my SIP Provider. My provider restricts me to send only 120 minutes call duration per Trunk per each day. This means around 30 calls per Trunk per day with an ACD of 3 minutes. Please help me how to configure this if it is possible? 2- I need to send P-Preferred-Identity<sip:xxxxxxxxx@domain.com> in SIP header for each Trunk while dialing out to provider. Please help how to configure this. Thanks in Advance. Rgrds-----------Abid SaleemTechnical Manager NGNTerminus Technologies

7 14

Please Help! CANCEL 481 Problem Call dons't hunup
by Leo 29 Jul '11

29 Jul '11

Hi: I have an issue with FS connected to a SIP gateway. After the Freeswitch update(FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500). When I receive a call from the SIP Gateway and try to hangup with "CANCEL" message the call before he call is answered, Our freeswitch replay SIP 481 Call Transaction does not exist. Please help! Leo Liu --------------------------------------------------------------- INVITE sip:9501351350010@38.106.1.72 SIP/2.0 Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236 From: "calling from" <sip:8675582928388@204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6 To: "9501351350010" <sip:9501351350010@38.106.1.72> Call-ID: ID236ThB-18461-236-2NeaL(a)204.110.11.29 Contact: <sip:8675582928388@204.110.11.29:5060> CSeq: 237 INVITE Max-Forwards: 70 User-Agent: ABC-Phone Supported: replaces Content-Type: application/sdp Content-Length: 141 v=0 o=- 34791986 84556339 IN IP4 204.110.11.29 s=SIP CALL c=IN IP4 204.110.11.29 t=0 0 m=audio 35974 RTP/AVP 18 a=rtpmap:18 G729/8000 ----------------------------------------------------------------------------------------------------- SIP/2.0 100 Trying Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236 From: "calling from" <sip:8675582928388@204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6 To: "9501351350010" <sip:9501351350010@38.106.1.72> Call-ID: ID236ThB-18461-236-2NeaL(a)204.110.11.29 CSeq: 237 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500 Content-Length: 0 ----------------------------------------------------------------------------------------------------- INVITE sip:88359501351350010@38.106.1.83 SIP/2.0 Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B Max-Forwards: 69 From: "calling from" <sip:8675582928388@38.106.1.72>;tag=9r2ScyaN12DpB To: <sip:88359501351350010@38.106.1.83> Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55 CSeq: 15529052 INVITE Contact: <sip:mod_sofia@38.106.1.72:5080> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 141 X-FS-Support: update_display Remote-Party-ID: "calling from" <sip:8675582928388@38.106.1.72>;party=calling;screen=yes;privacy=off v=0 o=- 34791986 84556339 IN IP4 204.110.11.29 s=SIP CALL c=IN IP4 204.110.11.29 t=0 0 m=audio 35974 RTP/AVP 18 a=rtpmap:18 G729/8000 ----------------------------------------------------------------------------------------------------- SIP/2.0 100 Trying Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B From: "calling from" <sip:8675582928388@38.106.1.72>;tag=9r2ScyaN12DpB To: <sip:88359501351350010@38.106.1.83>;tag=7DAE5EAC-1DD0 Date: Fri, 11 Aug 2000 05:54:04 GMT Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55 Server: Cisco-SIPGateway/IOS-12.x CSeq: 15529052 INVITE Allow-Events: telephone-event Content-Length: 0 ----------------------------------------------------------------------------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKBvFBUj7NNcQ1B From: "calling from" <sip:8675582928388@38.106.1.72>;tag=9r2ScyaN12DpB To: <sip:88359501351350010@38.106.1.83>;tag=7DAE5EAC-1DD0 Date: Fri, 11 Aug 2000 05:54:04 GMT Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55 Server: Cisco-SIPGateway/IOS-12.x CSeq: 15529052 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: <sip:88359501351350010@38.106.1.83:5060> Content-Type: application/sdp Content-Length: 200 v=0 o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83 s=SIP Call c=IN IP4 38.106.1.83 t=0 0 m=audio 17050 RTP/AVP 18 c=IN IP4 38.106.1.83 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes ACK sip:88359501351350010@38.106.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 38.106.1.72:5080;rport;branch=z9hG4bKc583vDrSjNDmQ Max-Forwards: 70 From: "calling from" <sip:8675582928388@38.106.1.72>;tag=9r2ScyaN12DpB To: <sip:88359501351350010@38.106.1.83>;tag=7DAE5EAC-1DD0 Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55 CSeq: 15529052 ACK Contact: <sip:mod_sofia@38.106.1.72:5080> Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236 From: "calling from" <sip:8675582928388@204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6 To: "9501351350010" <sip:9501351350010@38.106.1.72>;tag=ZD2XZveKXjHFc Call-ID: ID236ThB-18461-236-2NeaL(a)204.110.11.29 CSeq: 237 INVITE Contact: <sip:9501351350010@38.106.1.72:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 Remote-Party-ID: "Outbound Call" <sip:88359501351350010@38.106.1.72>;party=calling;privacy=off;screen=no v=0 o=CiscoSystemsSIP-GW-UserAgent 8984 5356 IN IP4 38.106.1.83 s=SIP Call c=IN IP4 38.106.1.83 t=0 0 m=audio 17050 RTP/AVP 18 c=IN IP4 38.106.1.83 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes ----------------------------------------------------------------------------------------------------- ACK sip:9501351350010@38.106.1.72 SIP/2.0 Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236 From: "calling from" <sip:8675582928388@204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6 To: "9501351350010" <sip:9501351350010@38.106.1.72>;tag=ZD2XZveKXjHFc Contact: <sip:8675582928388@204.110.11.29:5060> Call-ID: ID236ThB-18461-236-2NeaL(a)204.110.11.29 CSeq: 237 ACK Max-Forwards: 70 User-Agent: ABC-Phone Content-Length: 0 ----------------------------------------------------------------------------------------------------- CANCEL sip:9501351350010@38.106.1.72 SIP/2.0 Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236 From: "calling from" <sip:8675582928388@204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6 To: "9501351350010" <sip:9501351350010@38.106.1.72>;tag=ZD2XZveKXjHFc Call-ID: ID236ThB-18461-236-2NeaL(a)204.110.11.29 CSeq: 237 CANCEL Max-Forwards: 70 Content-Length: 0 ----------------------------------------------------------------------------------------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236 From: "calling from" <sip:8675582928388@204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6 To: "9501351350010" <sip:9501351350010@38.106.1.72>;tag=ZD2XZveKXjHFc Call-ID: ID236ThB-18461-236-2NeaL(a)204.110.11.29 CSeq: 237 CANCEL Content-Length: 0 ----------------------------------------------------------------------------------------------------- ACK sip:9501351350010@38.106.1.72 SIP/2.0 Via: SIP/2.0/UDP 204.110.11.29:5060;branch=ruan0236bKUBMUBqad02361311730236 From: "calling from" <sip:8675582928388@204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6 To: "9501351350010" <sip:9501351350010@38.106.1.72>;tag=ZD2XZveKXjHFc Contact: <sip:8675582928388@204.110.11.29:5060> Call-ID: ID236ThB-18461-236-2NeaL(a)204.110.11.29 CSeq: 237 ACK Max-Forwards: 70 User-Agent: ABC-Phone Content-Length: 0 ----------------------------------------------------------------------------------------------------- SIP/2.0 100 Trying Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77 From: <sip:88359501351350010@38.106.1.83>;tag=7DAE5EAC-1DD0 To: "calling from" <sip:8675582928388@38.106.1.72>;tag=9r2ScyaN12DpB Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55 CSeq: 101 BYE Timestamp: 965973271 0.000381 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500 Content-Length: 0 ----------------------------------------------------------------------------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 38.106.1.83:5060;branch=z9hG4bK2CE48C77 From: <sip:88359501351350010@38.106.1.83>;tag=7DAE5EAC-1DD0 To: "calling from" <sip:8675582928388@38.106.1.72>;tag=9r2ScyaN12DpB Call-ID: dc2ca10f-3292-122f-ceb5-000d56703e55 CSeq: 101 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ----------------------------------------------------------------------------------------------------- BYE sip:8675582928388@204.110.11.29:5060 SIP/2.0 Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c Max-Forwards: 70 From: "9501351350010" <sip:9501351350010@38.106.1.72>;tag=ZD2XZveKXjHFc To: "calling from" <sip:8675582928388@204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6 Call-ID: ID236ThB-18461-236-2NeaL(a)204.110.11.29 CSeq: 15529071 BYE Contact: <sip:9501351350010@38.106.1.72:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2932c1f 2011-07-14 00-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: Q.850;cause=102;text="RECOVERY_ON_TIMER_EXPIRE" Content-Length: 0 ----------------------------------------------------------------------------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 38.106.1.72;rport;branch=z9hG4bKSj5XgvcHUpm6c From: "9501351350010" <sip:9501351350010@38.106.1.72>;tag=ZD2XZveKXjHFc To: "calling from" <sip:8675582928388@204.110.11.29>;tag=s020ruan0236DvX9t53Sti02361311730236T6 Call-ID: ID236ThB-18461-236-2NeaL(a)204.110.11.29 CSeq: 15529071 BYE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 __________ Information from ESET NOD32 Antivirus, version of virus signature database 6335 (20110729) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

4 3

curl.run and curl parameter
by Daniel Knoll 29 Jul '11

29 Jul '11

Hey Freeswitch and Javascript Guys, i would like to use curl.run to post a file to googles speech api. How can i add curl parameters like "-F filename" to pass the filename. curl.run("POST", "https://www.google.com/speech-api/v1/", "recognize?xjerr=1&client=chromium&lang=de-DE&lm=builtin:dictation", my_callback, "my callback_arg\n",, Content-Type: audio/x-flac; rate=16000); Has anyone an idea ? Thanx a lot for answers. Daniel

2 1

DTMF echo/crossed?
by Gavin Henry 29 Jul '11

29 Jul '11

Hi all, We generally don't suffer any type of echo deploying customer multi-tenant systems, but an interesting one was reported by a customer who said they could hear DTMF on the line that they were not generating? This is part of a business centre multi-tenant setup with just one user on that domain who heard the DTMF. They are lots of domains on that system. Any suggestions? Thanks, Gavin. -- http://www.suretecsystems.com/services/openldap/ http://www.surevoip.co.uk

4 6

No audio, only on outgoing calls
by Erin O'Meara 29 Jul '11

29 Jul '11

I have a blue.box server setup with a cloudhost, setup IPKall & DIDforSale. Both incoming calls come thru to my phone connected to the blue.box, I setup outgoing calls with CallwithUs and a Toll Free termination provider. All outgoing calls from the phone thru the blue.box connect, but no audio in either direction. The phone is behind a PFSense Firewall/Router and I have a second phone connected to an Elastix PBX that I use for my day to day business and everything works as expected. I have disable the Block All firewall rule and replaced it with Pass All and am pretty sure at this point its not a Firewall issue. When I dial the extension's number, I can see voicemail wav files playing but again hear nothing. So from what I can determine this is a freeswitch to phone issue. I have chatted on IRC with people at #2600 and #Freeswitch and googled the symptoms and come up no solution. Hoping some has seen this before and has a solution so I don't have to smash anything in frustration. Regards, 206.905.9520 http://salmonbaytechnology.com

1 1

invoking php scripts from dialplan
by Sam 29 Jul '11

29 Jul '11

Hi, I wanted to know is there another method of invoking a php script from the dialplan besides using the following because I still cannot get the script to run for wahtever reason. Yes, I have opened up the ports on my firewall. When I run the php script directly from console it outputs fine, but i cannot get it to output via the dialplan.  <extension name="outbound-socket"> <condition field="destination_number" expression="^55(522)$"> <action application="set" data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/> <action application="socket" data="127.0.0.1:8021 async full"/> </condition> </extension> //I then ran fs_ivrd with this command: /usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8021 //my php script: ivrd-demo.php #!/usr/bin/php -q <?php // set a couple of things so we dont kill the system ob_implicit_flush(true); set_time_limit(30); // Open stdin so we can read the data in $in = fopen("php://stdin", "r"); // Connect echo "connect\n\n"; // Answer echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: answer\n\n"; // Play a prompt echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: playback\n"; echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n"; // Wait sleep(5); // Hangup echo "sendmsg\n"; echo "call-command: hangup\n\n"; fclose($in); ?>

2 1

Bad sound quality and Abnormally large timer gap 100953 detected!
by Joao Leme 29 Jul '11

29 Jul '11

I just set up Freeswitch Precompiled for Windows x64 ( http://files-sync.freeswitch.org/windows/installer/) on a VPS Windows Server 2008. Sound is really bad and I'm getting the following: 2011-07-27 12:33:15.012695 [WARNING] switch_time.c:244 Abnormally large timer gap 100953 detected! Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems. also 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 6 seconds 1 heartbeat (core) 2011-07-27 12:43:12.214843 [WARNING] switch_scheduler.c:114 Task was executed late by 4 seconds 2 check_ip (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 1 heartbeat (core) 2011-07-27 12:44:20.226562 [WARNING] switch_scheduler.c:114 Task was executed late by 8 seconds 2 check_ip (core) What can I do? Also, yesterday, when I first set it up, it was all good. Thanks, John

11 23

FS as registra server behind SBC
by Dome Charoenyost 29 Jul '11

29 Jul '11

Dear All, I found problem when setup FS behind Huwei SBC. my network diagram is. NAT Public IP NAT 10.x.x.x [SIP Client 1] ----- [ADSL Router] ---------------------------- SBC ---------------------------- FS / [SIP Client 2] ----- [ADSL Router] --------------------------/ It's work fine when call incomming and play sound. But i got 403 from SBC when i try to call Local extension or originate call to registerd user. I thing nat problem can someone show me sip profile config work in this case. BG Dome C.

2 2