June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Web page thoughts
by jcromes＠gmail.com 11 Jun '09

11 Jun '09

My company is currently investigating a couple of projects that may take me in the direction of FreeSwitch... In general, our management does not often consider open source software for projects such as this, but I've been successful in proving to them recently that open source can deliver. FreeSwitch is a *very* professional and polished product, I can tell - from the code and from the community. Unfortunately, I've been hesitant to send people to your webpage lately because it went downhill a few weeks ago. Whenever I think about one of our executives going to your webpage (after my recommendation) and seeing a picture of people clanking beer glasses, or some idiot tied up in phone cables, I cringe. I know you're advertising for ClueCon, but honestly, some of those huge images on your front page really knock your product down a peg in professionalism. Anyway, I'm pretty new to the community and I don't claim to be a web designer. You have an excellent piece of software, but if I didn't already know that about FreeSwitch, your webpage would not make a good first impression. Please take that for what it's worth... I wanted to voice my opinion because if I'm thinking it, others may be as well. Thoughts anyone? J (Have I mentioned how awesome your source browser is though??!!)

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Rejecting calls without answering
by Klaus Teller 11 Jun '09

11 Jun '09

Hi Team, I'm still in need of a way to reject a call without answering it. I very much appreciate your help. Klaus. -- GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02

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Help understanding DEBUG and INFO log
by Lars Zeb 11 Jun '09

11 Jun '09

In a dialplan, the action sets effective_caller_id_number to a value, however, in INFO, the displayed value is not the same as the set. Why? http://pastebin.freeswitch.org/9361 Thanks, Lars

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FIFO through javascript
by Baskar 11 Jun '09

11 Jun '09

Hi, I have configured inbound in FS SVN Trunk. i have written small program for inbound call to bridge. i have used session fifo session.execute( "fifo", "sales_fifo_1 out wait undef '/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav'" ); session.execute("bridge", "sofia/internal/1003%XXXXXXXXXXXX"); Inbound call pass through JavaScript session and play the voice file but it did not bridge to the extension 1003. It keep on playing the same voice file. how can i bridge the call after session.execute. session.execute( "fifo", "sales_fifo_1 out wait undef '/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav'" ); Can any one assist me to resolve the above problem -- Warm Regards, N.Baskar

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SIP Registration Address
by lee jason 11 Jun '09

11 Jun '09

Dear All, I just have a question, How can I use Freeswitch to blind two IP address for SIP registration at same port(5060 UDP)? Thanks a lot. Jason Lee >

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Finding all active calls belonging to the same phone
by John Wehle 11 Jun '09

11 Jun '09

To duplicate our old PBX park functionality I need for a user who's on a call to be able to pick up a second line and dial a number to park the other call which is on his phone. I have something working, however am curious if there's a better way to accomplish this. Specifically I'm curious if there's a recommended way to find all the calls to / from the same phone / channel. I ended up configuring *5 to call a javascript program which: a) Gets the uuid from the session and uses it to search "show channels" to find the channel name. b) Normalizes the channel name and uses it to search "show channels" to find an uuid associated with the channel which is different from the one that invoked *5. c) Uses the uuid from "b" to search "show calls" to find the peer uuid. d) Uses uuid_setvar to set hangup_after_bridge=false and uuid_transfer to transfer the peer uuid to the proper fifo. One of the problems I ran into is the channel name has slightly different formats depending on whether it is an inbound or outbound channel. E.g.: sofia/internal/1003(a)XXX.XXX.XXX.XXX sofia/internal/1003@XXX.XXX.XXX.XXX:5060 sofia/internal/sip:1003@YYY.YYY.YYY.YYY:5060;transport=udp;... where XXX is the freeswitch box and YYY is the phone. I created the following function to normalize the channel name for comparison: function normalize_channel_name (name, direction, ip_addr) { var re = /^sofia\//g; var length = name.search (re); var new_name = name; if (length == -1) return new_name; if (direction == "inbound") { re = /@.*$/g; new_name = name.replace (re, "@" + ip_addr); } else if (direction == "outbound") { re = /\/sip:(.*@[^:]*):.*$/g; new_name = name.replace (re, "/$1"); } return new_name; } Suggestions for a better approach? Keep in mind that my existing user population expects (for better or worse) to use *5 to park the call on their phone so I'm somewhat limited in what I can do. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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What's the right way to use skypiax with dialplan
by Jingwei Yang 11 Jun '09

11 Jun '09

Hi All, I just finished installing freeSwitch and Skypiax. And I'm able to use skype api directly via the sk command like the following: freeswitch(a)localhost.localdomain>sk console skypiax1 freeswitch(a)localhost.localdomain>sk CALL userAAA It works like a charm and userAAA is able to receive the call and answer it. However, I'm stuck in figuring out the right way to use Skypiax with a dialplan. I've put a dialplan like below into /usr/local/freeswitch/conf/dialplan/default.xml <extension name="skypiax"> <condition field="destination_number" expression="^2909$"> <action application="bridge" data="skypiax/ANY/userAAA"/> </condition> </extension> On the freeswitch console, I'm not sure how to trigger this dialplan correctly. I've tried something like freeswitch(a)localhost.localdomain>originate sofia/external/root(a)192.168.1.1002909 API CALL [originate(sofia/external/root(a)192.168.1.100 2909)] output: -ERR MANDATORY_IE_MISSING freeswitch(a)localhost.localdomain>originate sofia/external/localdomain@localhost 2909 API CALL [originate(sofia/external/localdomain@localhost 2909)] output: -ERR NORMAL_TEMPORARY_FAILURE All failed with errors indicated above. Please let me know what's the right way to originate the call. Thanks! Regards, -Jingwei p.s. my os is CentOS 5.3.

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palani vel sent you a Friend Request on Yaari
by palani vel 10 Jun '09

10 Jun '09

palani vel wants you to join Yaari! Is palani your friend? <a href="http://yaari.com/?controller=user&action=mailregister&friend=1&sign=YaariNY…">Yes, palani is my friend!</a> <a href="http://yaari.com/?controller=user&action=mailregister&friend=0&sign=YaariNY…">No, palani isn't my friend.</a> Please respond or palani may think you said no :( Thanks, The Yaari Team <font color="#505050">-----------------------------------------------------------</font> <font color="#808080">Yaari Inc., 358 Angier Ave NE Atlanta, GA 30312</font> <a href="http://yaari.com/?controller=privacy">Privacy Policy</a> | <a href="http://yaari.com/?controller=absn&action=addoptout&email=freeswitch-users@l…">Unsubscribe</a> | <a href="http://yaari.com/?controller=termsofservice&action=index">Terms of Service</a> YaariNYX927AYT966XXO323IQT265

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A Few New Blog Posts
by Michael Collins 10 Jun '09

10 Jun '09

FYI, I've added a few new posts on the main FreeSWITCH page: http://www.freeswitch.org/node/190 - OpenSimulator on EC2 http://www.freeswitch.org/node/191 - Rob Smart FS as-home-PBX (U.K.) how-to Enjoy! -MC

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Remove example.com gateway?
by Lars Zeb 10 Jun '09

10 Jun '09

Is it OK to remove the example.com gateway? I removed the example.xml files in sip_profiles/external and sip_profiles/internal and changed the default_provider from example.com to myprovider.com. But I still see myprovider.com as a gateway in sofia status. How do I get rid of this, of course, if it's OK. I want to simplify the configuration. I dialed 18885551212 and it went into [local.example.com] and 8885551212 it went into [eavesdrop]. I think I'm talking more than just the gateway, but I do want to make the dialplan work in the way I expect. Thanks, Lars

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