June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

CTI
by Maxim Tsvetov 29 Jun '09

29 Jun '09

Hello! We are seeking possibilities to use CTI features with Freeswitch. This features are: - click-to-dial - call popup - answer call,hangup - call transfer Does Freeswitch support any cti standarts (SIP CTI aka TR/87, TAPI, CSTA..) or there is already written module or third-party software? This solution should support 100-150 simultaneous сonnections from freeswitch users. Could you please share you experience with CTI. Regards, Maxim Tsvetov

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bridge call from outbound socket
by Nik Middleton 29 Jun '09

29 Jun '09

Hi Guys, Is it possible to bridge to another destination while controlling a call via the outbound socket? In other words, I'm controlling a call using an outbound socket and at some point want to originate a new call leg and bridge the two. If it can't be done that way, I'm thinking I could originate the call using an inbound socket, grab the uuid and then call " api uuid_bridge <uuid> <uuid>" ? Regards,

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ClueCon 2009 - Last Call For Early Birds!
by Michael Collins 29 Jun '09

29 Jun '09

Good news! It's not too late to sign up for the early bird special of $499 for ClueCon, but you'll need to act fast - July 1 is the last day for the early bird special and it's only a few days away. If you haven't already registered for ClueCon then please call us and do so right away. Expedia.com still has deals on hotel rooms but the Wyndham is filling up quickly so don't delay in getting your room booked. In other news we have a new media sponsor: Biz-News.com. They will be conducting interviews with some of the speakers this year and are helping to promote the event. Please drop by their site and check them out! Looking forward to seeing everybody this August! -Michael Collins http://www.cluecon.com 877.742.CLUE

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Re: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP?
by chschaef 29 Jun '09

29 Jun '09

>Hi, >there is a "edit online" option to use to change the option on the phone. >You have to first install SIP Voip Settings Tool from the Nokia forum, and >then change the option "Secure call" to prefered from NCT. >Ognjen Hello, did you actually try it and succeed? The description in the Nokia forum is existing but the implementation of the listener is apparently missing ... regards, chschaef -- View this message in context: http://www.nabble.com/Anybody-tried-Nokia-E71-%28symbian-S60-3rd%29-with-Pj… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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skypiax featur request
by Meftah Tayeb 29 Jun '09

29 Jun '09

hello, this message is for skypiax developers / contributor please i have to request: i know a software named "SiSky" this is a SIP to Skype software gateway that let you interconnect your SIP IpPbx to the Skype network if i dial a skype speed dial number from my IpPhone, skype will no to bi visible but skypiax: if i dial any number i get skype visible and i heare the ringing sound in skype cool you try to hide it in the next release? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4196 (20090629) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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Patton 4554 as gateway for a FreeSwitch installation
by Frederik Denkens 29 Jun '09

29 Jun '09

Hi, Following the recommendations of this list, we went with an external gateway to connect to the BRI based ISDN network. We'd like to configure a Patoon 4554 2xBRI <-> SIP gateway with Freeswitch to connect to the public ISDN network. The freeswitch config seems quite straightforward, but we don't manage to get the Patton to register with the Freeswitch and vise versa. Does anybody have some insight/sample config/tips they can share with us on this? Honesty obliges me to say that our experience with the Patton product is quite limited, which is not a big help for such a complex product. Many thanks in advance! Frederik.

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one-way audio after playback+bridge
by Artem Shiyanov 29 Jun '09

29 Jun '09

Hello! I got a problem with one way audio, symptoms are: firstly play audio file to channel A (A is hears sound) secondly bridge channel B with A (A doesn't hear B). Environment: - no NAT - User Agents being used X-Lite, EyeBeam, SJphone - same result for all of them- no audio, Wireshark shows that there is no RTP-flow to A from FreeSwitch - dialplan: <extension name="playback_media_file"> <condition field="destination_number" expression="playmedia"> <action application="answer"/> <action application="playback" data="test.wav"/> </condition> </extension> <extension name="Local_Extension_from_SP"> <condition field="destination_number" expression="^([0-9]{2,9})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> </condition> <condition field="${sip_to_host}" expression="^([^.]*)\..*$"> <action application="set" data="orgname=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <anti-action application="set" data="ringback=${us-ring}"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <anti-action application="answer"/> <anti-action application="export" data="sip_h_X-SPFrom=&quote;${sip_from_user}&quote;<${sip_from_uri}>"/> <anti-action application="export" data="sip_h_X-SPTo=<${sip_to_uri}>"/> <anti-action application="export" data="sip_h_X-SPCallId=${sip_call_id}"/> <anti-action application="bridge" data="sofia/external/${orgname}send2voicemail@ $${starpound_sip_app_server}"/> </condition> </extension> - Call routing scheme: user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc Exact description what's going on is: user A -> FS -(bridge)-> my B2BUA Then my B2BUA transfers (using re-INVITE's), on behalf of user, call to extension "playback_media_file" . After a while B2BUA transfer (re-Inviting) user to extension "Local_Extension_from_SP". This should create a new call to user B. As a result - A doesn't hear B, but B- is OK. On the contrary, if a call is routed (by B2BUA) to the "Local_Extension_from_SP" extension (ommiting "playback_media_file" ext) - everything is OK. What I've tried: - set parameter "inbound-proxy-media" to "true" in Sofia profile - set parameter "disable_rtp_auto_adjust to "true" in Sofia profile Nothing helps. Any help or thoughts would be MUCH appreciated! Artem

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hangup_cause NONE vs. NORMAL_CLEARING
by Nicolas Brenner 29 Jun '09

29 Jun '09

I have a small JS script that makes a call, plays a sound file and then hangs up. For each call it makes, I log the hangup_cause variable on the CHANNEL_HANGUP_COMPLETE event. Most of the time, when calls are successful, I get a NORMAL_CLEARING cause, but sometimes I'll get a NONE cause. I wanted to know what the difference between these two is, because there is no reference to NONE in the wiki (http://wiki.freeswitch.org/wiki/Hangup_causes ). Thanks, Nicolás

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Result of an application...
by Klaus Hochlehnert 29 Jun '09

29 Jun '09

Hi, maybe a stupid question, but how can I find out the result of an application? If I do (in perl) $session->execute("bridge", "user/${DestNr}\@$VARS{domain}"); How do I know if this was successful or if the user was busy or if the phone doesn't exist? Is there any status variable for the result of an execute (or even any other command)? Thanks, Klaus

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Talk to freeswitch from Adobe AIR application through XMLRPC
by Prabhuram Mohan 29 Jun '09

29 Jun '09

Hi All, Abode AIR based on flash player can be used to connect to freeswitch and issue commands through XMLRPC. I read the internet to do this plumbing and also got help through generous fellow developers from AIR & freeswitch community. Now that it is successfully done, here is the complete code for the benefit of people who are following suite. Comments are welcome!! Code available here - http://neoalchemist.tumblr.com/post/132134683 Thanks Prabhu

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