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Re: [Freeswitch-users] Inline execute (subject renamed)
by Michael Collins 16 Feb '12

16 Feb '12

Virendra, If inline execution is still confusing you then it's because you are probably having a difficult time visualizing the two steps in dialplan processing: parsing and executing. During the parsing phase, the dialplan module is going through each extension looking at the conditions. Each time a condition matches (shows "PASS" in the FS log) it then looks inside that condition to see if there are any actions. If there are actions inside the matching condition then each action is added to a "task list" in the order they appear in the dialplan. These actions are *NOT* executed right now. They are simply added to the big "to-do list" of actions. Then, once all the parsing is done, the dialplan module goes back to the task list and starts executing each action, one at a time. Does that part make sense? If not, then the following information won't help you... The "inline" execution of an action means that the action is executed *during the parsing phase*. In other words, when the dialplan module finds a matching condition and there is an action to add to the task list, it looks to see if there is an inline="true" attribute. If there is, then the dialplan parser basically says, "Oh, I should not add this action to the task list; instead I should *execute this action RIGHT NOW* before I continue." To see that for real, consider this extension: <extension name="inline test"> <condition field="destination_number" expression="^(5599)$"> <action application="log" data="INFO I'm first!"/> <action application="log" data="INFO Oh no you're not!" inline="true"/> <action application="log" data="INFO Bummer, dude!"/> </condition> </extension> So, when I dial 5599, which debug line gets printed first? To find out, check out this pastebin <http://pastebin.freeswitch.org/18406>. The dialplan processing starts at line #41 of the pastebin. Every log line that starts with "Dialplan:" represents the dialplan module parsing the dialplan. The interesting stuff occurs starting at line #269 - we match the condition on the extension. Line #270 is the first action from this condition to add to the task list. Now, look closely at lines #271-273. On line #271 the dialplan parser sees the action - but notice the "INLINE" there. At line #272 you see the actual log that the action is being executed. On line #273 is the actual log application's output. So, this action was not added to the task list but rather executed right there. It was "executed inline." Note that at line #274 it goes back to dialplan parsing and adding items to the task list. Skip down to line #294 - there you'll see the items from the task list actually being executed. The two log applications that were not executed "inline" finally get executed there at the end. If you really want to see how this works try adding and removing the inline="true" on the various actions in this extension. Watch the logs. You'll see that anything with inline="true" gets executed long before the items that don't have inline="true". Hope this helps. -MC On Tue, Feb 14, 2012 at 9:26 PM, virendra bhati <virbhati(a)gmail.com> wrote: > Hi, > > I have FS 1.0.6 "bridge" book and after reading chapters 5 and 8. Still > this is very confusing concept. that's why i want to know from you guys an > simple language with 2 or more example if you can reply with answer. Inline > execution is very confusing > concept for me. >

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NOTIFY not sent in multi-FreeSWITCH setup
by Vik Killa 15 Feb '12

15 Feb '12

I have 3 FS servers running together sharing UA registrations using ODBC Each FS server has a different "switch name": fs01, fs02, fs03 When a UA registers, it will randomly hit one of the 3 FS servers and "sofia status profile internal reg" will show something like: freeswitch@default> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 5e5392c0-932df231-3d1c57f2(a)192.168.99.69 User: 1004(a)test1.voip.demo.net Contact: "user" <sip:1004@72.43.207.130:5060;fs_path=%3Csip%3A72.43.207.158%3Blr%3Breceived%3Dsip%3A72.43.207.130%3A5060%3E> Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Status: Registered(UDP)(unknown) EXP(2012-02-15 09:45:47) EXPSECS(3449) Host: fs03 IP: 72.43.207.158 Port: 5060 Auth-User: 1004 Auth-Realm: test1.voip.demo.net MWI-Account: 1004(a)test1.voip.demo.net Call-ID: 93aa045c-fd5f2941-d207cb1e(a)192.168.99.21 User: 1000(a)test1.voip.demo.net Contact: "user" <sip:1000@72.43.207.130:1666;fs_path=%3Csip%3A72.43.207.158%3Blr%3Breceived%3Dsip%3A72.43.207.130%3A1666%3E> Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.2.0477 Status: Registered(UDP)(unknown) EXP(2012-02-15 09:29:41) EXPSECS(2483) Host: fs01 IP: 72.43.207.158 Port: 5060 Auth-User: 1000 Auth-Realm: test1.voip.demo.net MWI-Account: 1000(a)test1.voip.demo.net Notice the "Host" is the switch name.... Everything is working great except presence, it seems FS only sends a NOTIFY out when a call comes to the host that the UA is registered. So in this example, if UA 1004 gets a call on fs01 or fs02, a NOTIFY is not sent out, but if UA 1004 gets a call on fs03, the NOTIFY is sent out. Is there any way to remedy this?

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different moh in conference for each call?
by Anita Hall 15 Feb '12

15 Feb '12

Hi I am trying to play different moh sound for each conference call by passing a variable to moh-sound from the dialplan to autoload_configs/conference.conf.xml In my dialplan/default.xml <extension name="conference_incoming"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="knowlus_hold_music=$1" /> <action application="conference" data="3001@default"/> <action application="hangup" /> </condition> </extension> In my autoload_configs/conference.conf.xml <param name="moh-sound" value="local_stream://${knowlus_hold_music}"/> However, it does not work. It is not able to expand the variable ${knowlus_hold_music} In the log 2012-02-15 19:55:31.195264 [WARNING] mod_local_stream.c:393 Unknown source ${knowlus_hold_music}, trying 'default' Is it even possible to pass a channel variable to a config file ? Or is there another way to achieve what I am trying to do ? Thanks. regards, Anita

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Sip profile help
by Kevin Bushong 15 Feb '12

15 Feb '12

I'm new to Freeswitch so if this has been discussed, please send me to the proper link. I need to setup a trunk to Level3 (I've heard that's not popular at Freeswitch too :) ) Level3 does not use authentication but restricts based on IP address. When I setup the SIP profile, the username and password fields are required so I put in anything and set register to false. Whatever I put in the username field gets passed to Level3 as <username(a)xxx.xxx.xxx.xxx> in the FROM, Contact, and Remote-Party-ID fields. I've tried putting in nothing, but then I get <null(a)xxx.xxx.xxx.xxx> instead. How do set this to the E164 number of the calling party instead of the username?

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Alert... Brian West AKA BKW is back in the hospital
by Ken Rice 15 Feb '12

15 Feb '12

As you all know BKW one of the original members of the core FreeSWITCH developers team was in the hospital due to a mass in his chest... All was well until today and he's been re-admitted with some heart rhythm related issues... To keep Brian from constantly having to update people to his status, we have created a Jira to monitor his progress... http://jira.freeswitch.org/browse/BKW-1 Visit that ticket to get the latest updates. Also please save the get well wishes for the mailing list so that people can follow the ticket easily. Maybe we can get a patch to resolve this issue quickly! Brian Get Well Soon! Stay Tuned for More Info...

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FreeSWITCH Conference Call Today: 2600Hz Team
by Michael Collins 15 Feb '12

15 Feb '12

Hello all! Apologies for the late notice. When Brian West isn't at my beck and call I just can't function. :) Today's conference call will start shortly. Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_15 We are looking forward to having Darren Schreiber (IRC: pyite) and the rest of the 2600Hz team come talk to us about the state of their projects. -MC

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skipping vm and plays enter the pwd
by Aakash 15 Feb '12

15 Feb '12

Hi All, I have configured has any inbound calls enter to my inbound number play an ivr and transfer to user ext (eg 1080)pressed by customer,if the user dint pick the call enter to their voicemail automatically.But sometimes i am getting please enter password followed by # by skipping voicemail message.I have attached my logs below http://pastebin.freeswitch.org/18321 Regards, Aakash.V -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skipping-vm-and-plays-enter-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Tracking codec g729 by event
by Antonio 15 Feb '12

15 Feb '12

Hi, I'm trying to keep tracking on the use of codec g729 licenses on my server. I've a little ESL program that check all the events for a new call and if the call uses g729 i execute the api "g729_used" to fetch the value. This doesn't work, i try it on event CODEC, and EXECUTE. Is there any event that i can use that is trigger by the encoder/decoder in mod_com_g729 when it actives a license? Thanks, _________________________ António Silva

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FreeSWITCH Cookbook
by Craig Smith 15 Feb '12

15 Feb '12

When can we expect to be able to pick up a copy of the FreeSWITCH Cookbook?

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Choppy Audio Recordings
by aromberg＠gmail.com 15 Feb '12

15 Feb '12

Hello, I'm having a persistent problem across multiple installations. All are running GIT Head. Windows 2008 R2 x64 Dual Xeon 2.8, 4G of ram, Hyper-V instance, IIS running a few sites, nothing major. Debian 6, dual core 2.13GHz Xeon, 512mb dedicated 1gb burst, OpenVZ instance, freeswitch + small apache instance CentOS 5, 8 core 2.27 Xeon, 512MB RAM, Xen instance, freeswitch + apache (doing nothing) Centos6, single core 3.33Ghz 512MB ram, dedicated machine (no instance), nohz setting enabled per the wiki, only freeswitch running Any recordings come out choppy and slow, no matter what I do. Doesn't matter if it was WAV or MP3 using mod_shout. The endpoint I'm using is a Linksys SPA942. Any pointers would be helpful at this point. Thanks, Adam

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