- FreeSWITCH-users - FreeSWITCH Mailing Lists

Last Chance for ClueCon 2011: Get Your Hotel Today
by marketing＠cluecon.com 01 Aug '11

01 Aug '11

1 0

Still having problems making calls to phones behind double nat
by Rob Hutton 01 Aug '11

01 Aug '11

I am still unable to figure this one out and would appreciate any help. I have two phones at a remote location that can call out through the switch and have two way audio, but no one can call into them. The switch returns a 606 User Not registered even though show registrations seems to indicate that they are. I don't see anything in the sip trace to indicate that it actually tried to contact the destination user, but INFO keepalives are going back and forth... Configuration and sofia log are here: http://pastebin.freeswitch.org/16784

2 1

Re: [Freeswitch-users] 3
by Dave Bracken 01 Aug '11

01 Aug '11

Make the right choice.. http://kodaifresh.com/page.php?mgoogleId=69h8

2 1

unable to connect to freeswitch server using x-lite
by Dhairya Vora 01 Aug '11

01 Aug '11

I have installed freeswitch on centos ----------------------------------------------------- [root@localhost ~]# uname -r 2.6.18-238.19.1.el5 ------------------------------------------------------ I have installed x-lite on a different machine in same lan. when i try to connect x-lite to freeswitch with valid id, password, request times out. any idea to solve this? -- Dhairya Vora(newbie)

3 2

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