January 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Question about switch_cond_next()
by Alex To 29 Jan '10

29 Jan '10

Hi, would someone clarify this for me please Is it right to say that switch_cond_next() does only one thing which is to put the current thread to sleep for a period of time (1 second) ? Is it equivalent to, let's say, Thread.Sleep(xxx); or it also does something else besides putting the current thead to sleep ? Is switch_cond_yield the same as switch_cond_next, except that switch_cond_yield accepts parameter as the input interval ? Thanks Regards Alex To

2 2

[Freeswitch-dev] FreeSWITCH Conference Call Agenda
by Michael Collins 29 Jan '10

29 Jan '10

Greetings, This week's conference call agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_29 Please add your items as the agenda is very light this week. We do have a few things to discuss, though, so please hop on and bring a friend! Talk to you all soon, Michael

1 0

[Freeswitch-dev] .deb sound installation files
by Mark Sobkow 28 Jan '10

28 Jan '10

It looks like the sound files are no longer part of the standard build, as per emails that were sent out some time ago. Is there any word on .deb support for building the sound files? Our install process requires that we do .deb files for ubuntu rather than doing a source build. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

2 2

[Freeswitch-dev] fscomm Call failed with reason NO_ROUTE_DESTINATION.
by vip killa 27 Jan '10

27 Jan '10

Hi, I've downloaded and installed the latest FSComm for windows. I am able to register to the freeswitch server but i cannot make a call. When i try an extension i know works (because i use it with ekiga) i get "Call with (####) failed with reason NO_ROUTE_DESTINATION". I've tried calling ####(a)server.ip. But nothing works. I can receive calls to the FSComm application. I'm running FSComm on windows xp 64.

5 6

[Freeswitch-dev] FreeSWITCH 1.0.5 and Buying Dinner For The Dev Team
by Michael Collins 26 Jan '10

26 Jan '10

Hello all, The FreeSWITCH development team is planning to meet in one place during the week of February 8 to release version 1.0.5! We would like to invite everyone to show their appreciation by donating a few dollars to help pay for dinner for the development team. More information is available here: http://www.freeswitch.org/node/234 Let's all show the team how much we appreciate them by giving them a well-deserved dinner! -Michael

1 0

[Freeswitch-dev] Estimated release date for 1.0.5 final
by Roland Zagler 26 Jan '10

26 Jan '10

Hi everyone, does anyone already have an idea on when 1.0.5 final will be released? Or at least a timeframe... Thanks, Roland Zagler

4 3

[Freeswitch-dev] No sound with Gizmo5 after transfer from a dial plan
by Paul Li 24 Jan '10

24 Jan '10

It is a very interesting and challenging issue! What I am trying to achieve is playback a message (notice) to PSTN telephones with Gizmo5 as service provider with the originate command as follows: $ originate sofia/gateway/gizmo/0019876543210 5001 (Case 1) $ originate sofia/gateway/gizmo/0019876543210 5002 (Case 2) I can hear the playback from extension 5001 in the first case, while I could not hear anything from extension 5002 in the second case. The difference between them lies in that extension 5002 transfer the call when the PSTN phone is connected. Please see the dial plan below. Both dial plans are basically adapted from examples found in the online documents. Both 5001 and 5002 work for my two other voip service providers. On the surface, it appears to be Gizmo5's problem. But, I am wondering if there is any issue on the FreeSwitch side. My main goal is to delay playback until the end user has picked up the phone. I have been looking for a way to do this but ended up with nothing so far. I have tried setting {ignore_early_media=true}, but that did not help either. I am welcome any suggestion on how to achieve this. In the meanwhile, I would like point out this issue.  <extension name="direct_ivr_demo1"> <condition field="destination_number" expression="^5001$"> <action application="start_dtmf" /> <action application="playback" data="test.wav"/> </condition> </extension>   <extension name="indirect_ivr_demo2"> <condition field="destination_number" expression="^5002$"> <action application="set" data="execute_on_answer=transfer ANSWERCALL XML default"/> <action application="sleep" data="60000"/> <action application="transfer" data="NOANS XML default"/> </condition> </extension>  <extension name="Call_Answered"> <condition field="destination_number" expression="^ANSWERCALL$"> <action application="start_dtmf" /> <action application="playback" data="test.wav"/> </condition> </extension>  <extension name="Call_Not_Answered"> <condition field="destination_number" expression="^NOANS$"> <action application="hangup" data="NO_USER_RESPONSE"/> </condition> </extension>

2 4

[Freeswitch-dev] thesis on live sip calls migration
by stefano bossi 23 Jan '10

23 Jan '10

Hi all, as promised in #freeswitch-dev I'll try to explain better what I've done and I would like to achieve for my thesis. At the moment I successfully patched sofsip_cli and now it can do something near active call migration :) Thanks to the versatility of nua api I could achieve my objective with little effort. Basically I call someone, then I kill sofsip_cli and I restart it. I launch my "reinvite" command and the call can continue normally. The called client doesn't realize what happened. In this case I write call data in a file and when I launch the reinvite I read directly from this file. Sofsip_cli sends an INVITE message with some TAG appended to the nua_handle. So the called phone thinks to receive an in-session invite and simply refreshes the call, but Sofsip_cli instead allocates all the stuff for a new call. In this way I can avoid to modify directly the RTP part. It seems to be all simpler. I even wrote a little module for FS able to monitor(using the SIP OPTION message) the life of a specific sofia profile.. It's just a proof of concept but it can do failover(without live call migration) between 2 machines in about 50ms. This is possible with these actions: - registrations are shared with odbc - on the start of the module(present only on the backup machine): 1. I set an arp rule blocking the arp response for the virtual IP (set on the primary machine) 2. I set the virtual IP on the backup machine( but no one knows thanks to the arp rule and so I can bind to vIP) 3. I prepare and run the sip profile (on the backup)... loading here the profile permits to save a lot of time during reaction - on the reaction 1. I remove the arp rule 2. I send a gratuitous arp request As I said in about 50ms sip clients can call again. Maybe this time can decrease using NETLINK socket for the arp table. The union of these 2 works will give us a very fast "live profile migration" :D There some points to discuss: - switch_core_session_resurrect_channel: I didn't know this function, I need to understand what it offers, maybe tomorrow ;) - the propagation of call state variables: my first idea was to use the multicast events adding some headers to send all the necessary data (but anthm proposed XML) - I thought only to sofia aspects.. In next days I'll try to understand where to hook in FS to try these ideas. When I'll have something ready (and without very very big errors:) I'll be happy to send you the patch. This is my first work on something real like FS.. I'm opened to any kind of suggestions!* *please contact me for further clarifications Thanks Stefano Bossi

3 5

[Freeswitch-dev] FSComm Custom Parameters and Crashes
by Paul Li 23 Jan '10

23 Jan '10

Custom Parameter tab is having another problem and crashes under the following situation. If a gateway, gw1, has two custom params, and the second gateway, gw2, has zero or one custom param. After you first try to edit gw1, and then cancel to close, then you proceed to edit gw2, you will see blank row(s). When click OK, FSComm crashes, which might resulted from the blank row(s) in the table. João, it appears that you did not completely clears the entries for the Advance (custom) tab. Other than this problem, everything else looks good, thanks for your hard work. Next time, I will try to log any issue with Jira.

2 1

[Freeswitch-dev] FSComm preferences
by Dmitry M 23 Jan '10

23 Jan '10

Checking phones preferences I noticed Caller id Name and Number located in General Preferences. IMO it'll be more natural to move them into Account. I expect account may have different proto (SIP,IAX,H323,...), different caller id, different codec settings, etc. Dmitry

2 1