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RTMP buffer size?
by peely 08 Jul '11

08 Jul '11

Hi, I've been having a good old play with mod_rtmp, which is a very exciting addition, thanks! The main feedback I get when testing with colleagues is the latency, which seems to be somewhere in the order of 1.5 seconds each way. My testbed is an RTMP client > FS server > PSTN so I could three points where jitter buffers and encoders would be maintained. Ping time between servers is in the order of 50-60ms. Is there a way of controlling the size of the jitter buffer in mod_rtmp to reduce it? jitter_buffer_msec doesn't seem to change anything in this scenario. On another note, it seems the option is rtmp.conf.xml to allow calls without rgistration doesn't seem to enable this option, I always seem to have to get the client to log in. Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RTMP-buffer-size-tp6554200p65… Sent from the freeswitch-users mailing list archive at Nabble.com.

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New BSD timer module
by Steven Ayre 08 Jul '11

08 Jul '11

Hi everyone, I've submitted a Jira with a proposed new mod_kqueue timer module. This is a BSD equivalent of mod_timerfd. http://jira.freeswitch.org/browse/FS-3398 I'd appreciate it if anyone that uses FreeSWITCH on BSD that feels like it would give it a quick test to see how well it works. To install it: - Create a new directory src/mod/timers/mod_kqueue - Place the mod_kqueue.c file from the jira in that directory - Add timers/mod_kqueue to modules.conf - Build and install - Load it from modules.conf.xml or fs_cli. Regards, Steven Ayre

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Lua to use mod_lcr's "as xml"
by Avi Marcus 08 Jul '11

08 Jul '11

How can I run mod_lcr's "as xml" and then use lua to deal with the result as an array or table? Thanks, Avi

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Transcoder cards & g.729 licenses
by Nandy Dagondon 08 Jul '11

08 Jul '11

hello guys, if i use a transcoder card e.g. sangoma D150 for TDM-to-VoIP setup, do i still need to purchase 30 G.729 licenses for every E1 trunk? if yes, will the licenses be bound to the transcoder card? tks. -nandy

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Re: [Freeswitch-users] From header rewrite on B2BUA scenario
by Peter Olsson 08 Jul '11

08 Jul '11

Make sure to set caller-id-in-from to true in the gateway config. /Peter ----- Reply message ----- Från: "Dave Horton" <dave(a)dchorton.com> Datum: fre, jul 8, 2011 06:53 Rubrik: [Freeswitch-users] From header rewrite on B2BUA scenario Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> I have a scenario where I have FS configured so as to take an incoming call and send it out through a configured gateway. Now this gateway does not require authentication, so in my gateway configuration I have the register param set to false: <gateway name="gwInternal"> <param name="username" value="foo"/> <param name="password" value="bar"/> <param name="realm" value="204.215.65.202"/> <param name="proxy" value="204.215.65.202"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> Now, even though I am not registering to this gateway, FS requires me to provide a username and password or else I get an error when starting/loading this configuration. So you can see above that I have just put in a username of "foo", just to have something there. Now I come to the problem. The result of the above configuration is that the From header coming in looks like this: From: <sip:+15083084809@204.215.65.201:5060>;tag=3519088051-756100 and the one that FS creates to send out through this gateway looks like this: From: "+15083084809" <sip:foo@204.215.65.202;transport=udp>;tag=Sr6taB8rFgt1H You can see that it has inserted the username of 'foo' in the user part of the uri of the From header. The reason that this is a problem is that this INVITE is going out to another remote FS, as it turns out. And when the dialed remote extension off that FS PBX does not pick up it then goes into voicemail, and the username of 'foo' is what is stored as the calling number. So when the user on that remote PBX collects that voicemail and tries to do call return, the FS ends up trying to transfer a call to 'foo', which fails. So call return is not working in the remote FS due to this rewriting of the From header. Is there a way I can configure my FS to send calls out through a gateway and maintain the user part of the From URI that was received on the incoming call? Or, alternatively, can I configure the voicemail application to look for the caller id field from somewhere else than the user part of the From URI ? Dave _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4e168bec32761053165832!

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From header rewrite on B2BUA scenario
by Dave Horton 08 Jul '11

08 Jul '11

I have a scenario where I have FS configured so as to take an incoming call and send it out through a configured gateway. Now this gateway does not require authentication, so in my gateway configuration I have the register param set to false: <gateway name="gwInternal"> <param name="username" value="foo"/> <param name="password" value="bar"/> <param name="realm" value="204.215.65.202"/> <param name="proxy" value="204.215.65.202"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> Now, even though I am not registering to this gateway, FS requires me to provide a username and password or else I get an error when starting/loading this configuration. So you can see above that I have just put in a username of "foo", just to have something there. Now I come to the problem. The result of the above configuration is that the From header coming in looks like this: From: <sip:+15083084809@204.215.65.201:5060>;tag=3519088051-756100 and the one that FS creates to send out through this gateway looks like this: From: "+15083084809" <sip:foo@204.215.65.202;transport=udp>;tag=Sr6taB8rFgt1H You can see that it has inserted the username of 'foo' in the user part of the uri of the From header. The reason that this is a problem is that this INVITE is going out to another remote FS, as it turns out. And when the dialed remote extension off that FS PBX does not pick up it then goes into voicemail, and the username of 'foo' is what is stored as the calling number. So when the user on that remote PBX collects that voicemail and tries to do call return, the FS ends up trying to transfer a call to 'foo', which fails. So call return is not working in the remote FS due to this rewriting of the From header. Is there a way I can configure my FS to send calls out through a gateway and maintain the user part of the From URI that was received on the incoming call? Or, alternatively, can I configure the voicemail application to look for the caller id field from somewhere else than the user part of the From URI ? Dave

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Playing multiple files simultaneously
by Bryan Smart 07 Jul '11

07 Jul '11

Is it possible for Freeswitch to play more than one file to a channel at a time? What I've seen and tried from the dialplan and scripts either queues files to play, or will stop a currently playing file so that the newly requested file will play. This also seems to be the case in conferences. When I send multiple play commands to conferences, the files are queued. As for how this might be used, think of an IVR that plays queued prompts, yet continuously plays looping music or a Shoutcast stream in the background. I also want to be able to play short cue tones that start at the same time as a prompt (don't want to pre-mix them in to a single file, though). Is this currently possible through any means? Perhaps with the event socket? Bryan

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Freeswitch dynamic routing of all calls
by Mateusz Bartczak 07 Jul '11

07 Jul '11

Hi all I'm new to FS and I would like to know is it possible to implement following scenario: 1. User dials number 2. Routing script detects outgoing call event. Every call needs to be handled by routing script. 3. Routing script takes in input: user name, domain, dialed number. Than it query database to find best SIP trunk to route the call, it also checks destination price per minute and calculates maximum call time for prepaid user. 4. Routing script output is: SIP trunk to use, SIP call parameters (ie. callerid), maximum call duration 5. FS read output from routing script and make call using returned parameters Preferred routing implementation technology: background running unix deamon written in Java or PHP. Connection with FS via socket. Event routing script will be multi-threaded, must be able to deal with a lot of calls in parallel and processing of one call should not block processing of other calls (I have this problem with Yate voip server, and that's really big problem) Is it possible to do this using FS? Any advices where to search for additional info? I know that there is event handler but can it return "dialstring" for outgoing call events? Some code examples? I will really appreciate your help

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Having problem completing calls between two phones behind double NAT...
by Rob Hutton 07 Jul '11

07 Jul '11

The network layout it something like: FS and Ext 3 <- Local Net ->Inet Rtr(Static NAT) <== INET ==>Inet Rtr(Dynamic NAT)<- Local Net -> Ext 1 and Ext 2 where: The public IP of the Freeswitch box is static There are 2 sip profiles, one for the internal clients on 5060 and one for the NATd clients on 5070 with the external IP hard set and Freeswitch running with -nonat Ext 3 is on the same subnet as the FS server and is using sipinterface_1 to connect Ext 1 and Ext2 are on the same subnet on the remote side of the double nat and using sipinterfacec_2 I can call from Ext 1 to Ext 3 with no problem I can call from Ext 2 to Ext 3 with no problem When I try to call from Ext 1 to Ext 2 or Ext 2 to Ext 1, I get a 606 USER_NOT_REGISTERED show registrations seem to indicate that they are, and I can't find any difference in the call processing to the extensions that it works to, and those that it doesn't. Pastebin with registrations and failed call log is: http://pastebin.freeswitch.org/16705 Thanks in advance for any help...

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absolute_codec_string
by David Villasmil 07 Jul '11

07 Jul '11

Hello Guys, I might be getting it wrong, but i'm trying to send out for the B-leg ONLY g.729 with this in the dial plan: <variable name="absolute_codec_string" value="G729"/> FS is still offering what it gets on the A-leg... any ideas? Thanks David

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