April 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Determine IP address of sender
by Alois Komenda 09 Apr '08

09 Apr '08

Hello, is there a way in FS to determine the IP address of the sender of a SIP message. I'm not talking about IP addresses or domain names that are sent on application layer (i. e. within SIP) but on network layer. I need this to determine if one network element pretends to have several (SIP-)identities. Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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mod_xml_curl and user directory
by Jonas Gauffin 09 Apr '08

09 Apr '08

Hello som phones have different fields for auth username and username. Freeswitch sends the auth-username when requesting the directory, but then registers the user using the regular username. For me this is a problem, since I generate dialplans dynamically in my app using the info sent in the fetch directory request. Is it possible that FS can send both names when it requests the directory for a user? Thanks, Jonas

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Problem playing media
by Nicolas Brenner 09 Apr '08

09 Apr '08

Hello everyone, I'm having some trouble with FS :( apparently with mod_shout. I want to play an mp3 file after answering a call so I compiled mod_shout following the wiki, then configured an extension to answer a call and play an mp3 file I uploaded to the server. The thing is, FS supposedly plays the file, but I can't hear it on the softphone with which I'm calling, and also, after playing the file, FS seems to freeze and then I get a lot of: sofia_event_callback() event [nua_r_bye] status [408][Request Timeout] session: n/a sofia_event_callback() event [nua_i_state] status [408][to BYE] session: n/a lines on the console/log. Also, after I try to call once, the softphone does not work anymore, and I have to make it register again with FS. After all this, when I shutdown FS, it takes some time, and while it's trying to shutdown it prints a lot of the same lines as above. I thought it was mod_shout, so I commented it from autoload_configs/modules.conf.xml so it does not load, and replaced the mp3 file with a wav file (FS has permission to read both of them), but I'm getting the same behaviour. Now I'm even getting that behaviour when dialing 9999 or 9998 (default dialplan pre-configured extensions). Now I can't get FS to work right, everytime I try to make a call, either initiating it from the console, the softphone or js, I get the same weird behaviour. Any clues? I'm sorry if this is trivial or a known issue, but I haven't been able to figure it out, thanks a lot for your time and help. -- Nicolás Brenner

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New to FreeSwitch
by Ritesh Singh 09 Apr '08

09 Apr '08

Hi All, I am very new to freeswitch. It will be great if some one can tell me few things: 1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y. Then what changes i am supposed to do and at what place. I would like to use the mod_portaudio for this purpose. 2) I started the windows freeswitch and loaded port_audio by using "load mod_portaudio" , then used "pa call 1234"...the session gets initiated but the call gets hanged up with the log "portaudio/1234 [CS_RING] [NO_ROUTE_DESTINATION]". 3) Suppose i have a jabber server x.net. and i have 2 users having account at that jabber server. Then how can i use mod_dingaling so that i have the voice chat between those user of the jabber server x.net. Also, i would like to dump the all the record regarding the call made, like duration of call , person who initiated the call, call destination , etc. Any help in this regard is highly appreciable. Please do forgive me if you think its stupid mail but i am desparate for these answers. This is my second mail as no one replied my first mail. Please do reply this mail else i ll be broken .... Thanks and Regards Ritesh

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great review of Freeswitch on ZDNet
by Brian Burt 09 Apr '08

09 Apr '08

*Open source projects have revolutionized operating systems, web servers, web browsers, and so why not carrier switches? The FreeSwitch open source project <http://www.freeswitch.org/> released its Release Candidate 1 (RC1) yesterday providing and by early accounts the software rocks.* http://blogs.zdnet.com/Greenfield/?p=214#comments Congratulations! BTW, we're planning to use it here to create flexible subconferences. Participants can be in partners but everyone hears the moderator with instructions ... then move into 4s or 6s for small group discussion, raise their hand with a pound key if someone has a question for an assistant, etc. Please let me know if you're interested in assisting with the development of that! I think it could really revolutionize "teleconferences" which are currently passive/snoozeville. When people are in "listen only" they might as well be sleeping when it comes to comprehension/retention. All the best, -Brian -- ===================== Brian Burt, Founding Partner Financial Circles Empowerment in Community www.FinancialCircles.com <http://www.financialcircles.com/> 415-308-4258 mobile (best) 510-251-1189 office info(a)FinancialCircles.com ===================== - - - "Financial Circles teaches people to master money so that it serves their dreams. We offer courses and an active network where people wake up to possibility, master concrete financial and business skills, and take action on what truly matters to them." - -

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[Freeswitch-users] http://digg.com/software/An_Open_Source_Class_Five_Telephony_Switch
by Ken Rice 08 Apr '08

08 Apr '08

Hey Guys.. Its getting Dugg... Join in and get the word out http://digg.com/software/An_Open_Source_Class_Five_Telephony_Switch

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ivr scripts and streamfile
by Jonas Gauffin 08 Apr '08

08 Apr '08

Hello Sometimes the sound files are not said properly. This usually happens for phrases that are said directly after session.answer, or after a "lenghty" function (as fetching xml using curl). For instance in my voicemail I have a phrase saying "You have no new messages" directly after session.answer. Sometimes I just hear "no new messages" when dialing in. What should I do? Thanks, Jonas

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preAnswer : API x JS
by Daniel Eli 08 Apr '08

08 Apr '08

Hi, first of all, congratulations to the FS team. I am trying to control FreeSwitch with a telnet client using the API. I can't find some way to pre-Answer (code 180) a channel in this interface, like the preAnswer method found in the session object. Another idea is calling an js script with the API, for example: bgapi jsrun preanswr.js ..., but there is another problem: the default session object is not created and I can't find how to get the session object from the uid of the channel (or from another channel variable). So, is there any way to get one session object from some channel variable ? Or, is there any way to pre-answer (generate the 180 code) the channel with the API ? Could you help me about that? Thanks in advance. Daniel. ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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ENUM lookup and simultaneous dialing
by Mikael A. Bjerkeland 08 Apr '08

08 Apr '08

When doing an enum lookup where a number has two NAPTR records with the same preference and order enum_auto_route gets set to: sofia/gateway/openser/bordtelefon(a)192.168.1.154| sofia/gateway/openser/mikael(a)some-gw.com The "|" sets the dialstring for chained dialing. If the order and pref is identical for both records, shouldn't the dialstring be using "," to call both simultaneously? Is this a feature or a bug? Regards, Mikael

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About Dingaling
by Ali Jawad 08 Apr '08

08 Apr '08

Hi All I am trying to test the jingle features of FS. I have compiled and setup FS on debian all went well and I made a call using softphones from 1000 to 1001. After that I read http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working I did the changes and rebuilt I edited my config files my ./conf/jingle_profiles/client.xml looks as follows <include>  <x-profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="thisisme(a)gmail.com/talk"/> <param name="password" value="thisIsMyPwd"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/>  <param name="auto-login" value="true"/> <param name="auto-reply" value="Press *Call* to join my conference"/> <param name="exten" value="1000"/> <param name="ext-rtp-ip" value="stun:stun.xten.com"/>  <param name="sasl" value="plain"/>    <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/>  <param name="exten" value="888"/>    <param name="vad" value="both"/>  </x-profile> </include> As you can see above I used <param name="exten" value="1000"/> after that I logged in with my SIP softphone "Eyebeam" and tried to make a call to gmail account that did not work..but I was able to make a conference call to conf+888(a)conference.freeswitch.org The server is running on a PUBLIC IP So what must I do/change/add/check to able to make calls to my gmail accounts (or from my gmail account to my sip phone) as mentioned in the FAQ. And where can I debug this issue, if I call 9999 I can see the call setup and successful call in my putty shell, however when I call myaccount(a)gmail.com nothing happens. Thx Q: What? Did you say it can talk to GoogleTalk? Yes in March of 2006 I developed my own XMPP telephony signaling library that is capable of communicating with Google's GoogleTalk. With a single Jabber account you can receive endless simultaneous calls from GoogleTalk clients and gateway those calls to IVR or another voice protocol like SIP or H.323. When FreeSWITCH is on both ends of the call you can bypass NAT and send extended data such as Caller ID and DNIS. With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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