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No Voice
by M.Emran 26 Aug '08

26 Aug '08

i have installed FS in public.And configured profile as follows: <profile name="doublenat"> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="6000"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="force-register-domain" value="$${domain}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> my dialplan is like this : <extension name="echo test"> <condition field="destination_number" expression="999"> <action application="answer" /> <action application="echo"/> </condition> </extension> <extension name="curl_test"> <condition field="destination_number" expression="(.*)"> <action application="set" data="proxy_media=true"/> <action application="bridge" data="sofia/internal/$1%$${domain}" /> <action application="hangup" /> </condition> </extension> I have called from my pc to another pc with X-Lite. I can not hear voice from any end. i have dialed 999 for echo test.that also not working. any help will be appreciatable. -- Regards ---------- M Emran InSpiration Software Ltd. Web: www.inspiresoftbd.com

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script finished before call is answered
by Yanhao Zhu 26 Aug '08

26 Aug '08

Hi, all, I have a very simple lua script hello.lua: session:answer() session:streamFile("hello.wav") session:hangup() When I tried "originate sofia/default/1001 &lua(hello.lua)", the script started to execute after the call is answered. But when I tried with something like "originate sofia/gateway/onesuite/123456789 &lua(hello.lua)", the script executed/finished before the call was answered. Is there a way to make sure the script only executes after the call is answered (by the end device) when calling through a SIP gateway? Can any one give me some hints? Thanks a lot, Yanhao Zhu

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wiki att_xfer or consultation hold ?
by Lee JJ 26 Aug '08

26 Aug '08

Hello : http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer I found this example seems not an "attended transfer" , rather a consultation hold. An ext 1527 --> call ext 1528 , then 1528 press "*3" , and keyin "1529#" 1527 into holding state listening music, 1528 and 1529 bridged if 1527 hungup , all disconected , only on 1529 hangup , 1527 go back with 1528 >From the following ftp://ftp.rfc-editor.org/in-notes/internet-drafts/draft-ietf-sipping-servic… 2.2. Consultation Hold Alice Proxy Bob Carol | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | |------------->| | |(100 Trying) F3 | | | |<---------------|180 Ringing F4| | | |<-------------| | | 180 Ringing F5 | | | |<---------------| 200 OK F6 | | | |<-------------| | | 200 OK F7 | | | |<---------------| | | | ACK F8 | | | |--------------->| ACK F9 | | | |------------->| | | Both way RTP Established | | |<=============================>| | | |INVITE(hold) F10 | |INVITE(hold) F11|<-------------| | |<---------------| | | | 200 OK F12 | | | |--------------->| 200 OK F13 | | | |------------->| | | | ACK F14 | | | |<-------------| | | ACK F15 | | | |<---------------| | | | No RTP Sent! | | | | INVITE F16 | | | |<-------------| | | | | INVITE F17 | | |--------------------------------->| | |(100 Trying) F18 | | |------------->| | | | | 180 Ringing F19 | | |<---------------------------------| | | 180 Ringing F20 | | |------------->| | | | | 200 OK F21 | | |<---------------------------------| | | 200 OK F22 | | | |------------->| | | | ACK F23 | | | |<-------------| | | | | ACK F24 | | |--------------------------------->| | | Both way RTP Established | | | |<=================>| | | BYE F25 | | | |<-------------| | | | | BYE F26 | | |--------------------------------->| | | | 200 OK F27 | | |<---------------------------------| | | 200 OK F28 | | | |------------->| | | | INVITE F29 | | | INVITE F30 |<-------------| | |<---------------| | | | 200 OK F31 | | | |--------------->| 200 OK F32 | | | |------------->| | | | ACK F33 | | | |<-------------| | | ACK F34 | | | |<---------------| | | | Both way RTP Established | | |<=============================>| | | BYE F35 | | | |--------------->| BYE F36 | | | |------------->| | | | 200 OK F37 | | | |<-------------| | | 200 OK F38 | | | |<---------------| | | | | | | In this scenario, Alice calls Bob. Bob places call on hold. Bob calls Carol. Bob then disconnects with Carol, then takes the call with Alice off hold. The call ends when Alice hangs up. 2.5. Transfer - Attended Alice Bob Carol | | | | INVITE F1 | | |--------------->| | | 180 Ringing F2 | | |<---------------| | | 200 OK F3 | | |<---------------| | | ACK F4 | | |--------------->| | | RTP | | |<==============>| | |INVITE (hold) F5| | |<---------------| | | 200 OK F6 | | |--------------->| | | ACK F7 | | |<---------------| | | No RTP | | | | INVITE F8 | | |------------->| | | 180 Ringing F9 | |<-------------| | | 200 OK F10 | | |<-------------| | | ACK F11 | | |------------->| | | RTP | | |<============>| | |INVITE (hold) F12 | |------------->| | | 200 OK F13 | | |<-------------| | | ACK F14 | | |------------->| | | No RTP | | REFER Refer-To: C F15 | |<---------------| | |202 Accepted F16| | |--------------->| | | NOTIFY F17 | | |--------------->| | | 200 OK F18 | | |<---------------| | | INVITE Replaces: B F19 | |------------------------------>| | 200 OK F20 | |<------------------------------| | ACK F21 | |------------------------------>| | RTP | |<=============================>| | | BYE F22 | | |<-------------| | | 200 OK F23 | | |------------->| | NOTIFY F24 | | |--------------->| | | 200 OK F25 | | |<---------------| | | BYE F26 | | |<---------------| | | 200 OK F27 | | |--------------->| | In this scenario, Alice calls Bob. Bob puts Alice on hold then calls Carol to announce transfer, then places Carol on hold. Bob transfers Alice to Carol which replaces the session between Bob and Carol. thanks!

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Can we have a forum
by Lee JJ 25 Aug '08

25 Aug '08

Dear Sirs : Can we have a forum like phpbb or punbb . thanks

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Project "Commercial PBX using FreeSWITCH", question 3
by Jon Bruel 25 Aug '08

25 Aug '08

I have a dream, said M. L. King. And so do I: To create a commercial alternative to Avaya, Cisco BroadSoft and others based on open source components. At this stage, I'm investigating different possibilities including the FreeSWITCH platform before a prospect is with the aim of raising 2-5 mio. USD for the initial steps to create ready-to-go-IP-PBX-package for big enterprises and telcos. I have a number of questions regarding the possibilities of the FreeSWITCH, and I have chosen to use this community to share the questions. Question 3: My experience with Asterisk has proven that the ability to monitor DTMF after calls have been bridges is very useful. With this facility you can transfer a call by pressing DTMF # or another predefined digit from any type of phone. Is this possible with FreeSWITCH? And how? Further if it is possible, what is the cost in terms of CPU? Thanks Jon Consiglia Telecommunications Hulsøvang 16 2960 Rungsted Kyst Denmark 45 16 1000 CVR: 27 04 78 82 www.consiglia.dk

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Project "Commercial PBX using FreeSWITCH", question 2
by Jon Bruel 25 Aug '08

25 Aug '08

I have a dream, said M. L. King. And so do I: To create a commercial alternative to Avaya, Cisco BroadSoft and others based on open source components. At this stage, I'm investigating different possibilities including the FreeSWITCH platform before a prospect is with the aim of raising 2-5 mio. USD for the initial steps to create ready-to-go-IP-PBX-package for big enterprises and telcos. I have a number of questions regarding the possibilities of the FreeSWITCH, and I have chosen to use this community to share the questions. Question 2: What are the possibilities to establish carrier grade system using the FreeSWITCH? In practice this requirement will need a setup, where there is a redundant server, which takes over, if the prime server fails, including all the management software which reacts correctly on failure signals from the prime server! Thanks Jon Consiglia Telecommunications Hulsøvang 16 2960 Rungsted Kyst 45 16 1000 CVR: 27 04 78 82 www.consiglia.dk

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Project "Commercial PBX using FreeSWITCH", question 1
by Jon Bruel 25 Aug '08

25 Aug '08

I have a dream, said M. L. King. And so do I: To create a commercial alternative to Avaya, Cisco BroadSoft and others based on open source components. At this stage, I'm investigating different possibilities including the FreeSWITCH platform before a prospect is with the aim of raising 2-5 mio. USD for the initial steps to create ready-to-go-IP-PBX-package for big enterprises and telcos. I have a number of questions regarding the possibilities of the FreeSWITCH, and I have chosen to use this community to share the questions. Question 1: Well, as this project is likely to be quite expensive, why create needles competition? So if anybody is working on a similar project it might be better to cooperate than compete! So please let me hear if you know about any similar project! Thanks Jon Consiglia Telekommunikation Hulsøvang 16 2960 Rungsted Kyst 45 16 1000 CVR: 27 04 78 82 www.consiglia.dk

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Please don't Hijack Threads.
by Brian West 25 Aug '08

25 Aug '08

As a note... If you send an email to the mailing list DO NOT CLICK REPLY and erase the body and change the subject. Doing this causes your email to hijack a thread. Please start a FRESH email with your questions or problems. Thread hijacking is bad. Thanks, Brian West sip:brian@freeswitch.org

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FreeSwitch dies when calling sys.exit() from hangupHook function in Python.
by Marc Orenberg 25 Aug '08

25 Aug '08

When I use the Python function sys.exit() in the hangup callback function, it immediately causes the FreeSwitch application to stop running. But I need to somehow stop the Python script from running in this situation. Here is a simple example: def hangup_hook(session, what): print("info","Hangup hook for %s!!\n\n" % what) sys.exit() def handler(session, args): session.answer() session.setHangupHook(hangup_hook) while(1): session.streamFile(PROMPTS_DIR + "ThanksForCalling.wav") The above script simply plays "Thanks for calling" over and over. If I hang-up, FreeSwitch dies. If I remove the sys.exit(), it will return from hangup_hook and continue trying to play the prompt forever. Thanks for your help.

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Re: [Freeswitch-users] SBC Abilities
by eric c 25 Aug '08

25 Aug '08

> We have global rate limit and ACL but not rate limit per IP. > We don't have any automatic blacklist but it's a good idea. That would definitely be a good idea; especially for not necessarily malicious users; what about bad IADs in the field. Also, testing out things with rfc4475 (SIP Torture Test) http://tools.ietf.org/html/rfc4475 Also, any testing with tools such as: voiper (http://voiper.sourceforge.net/) INVITE Flooder (http://www.hackingvoip.com/sec_tools.html) SipBomber (http://www.metalinkltd.com/downloads.php) I understand that people are placing FS on a protected network but if it's going to face the internet as an SBC, it should at least be fortified to handle the beating. thanks for your time and great work, eric clope

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