Re: [Freeswitch-users] FreeSwitch setup as a Dumb SBC

2009-02-05 Thread rod
Hi, how many static xml files did you create for your test ? rod. kokoska rokoska wrote: Anthony Minessale napsal(a): What does it look like if you serve the directory from the static xml file out of curiosity. Well, I write all user infos into static xml files loaded at startup

Re: [Freeswitch-users] origainate through sofia gateway

2009-02-05 Thread Jacek Sokulski
the result of apiExecute(bgapi, originate {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1...@192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline); /: 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 switch_ivr_originate()

Re: [Freeswitch-users] Freeswitch freezes on increasing call traffic

2009-02-05 Thread shehzad p
Hi Brian, As it can be seen from the system information, there require any change in system or any suggestion... out put of uname -a is : Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT 2007 i686 i686 i386 GNU/Linux Thanks, msp shehzad p wrote: HI Brian,

Re: [Freeswitch-users] Freeswitch freezes on increasing call traffic

2009-02-05 Thread Anthony Minessale
First of all please stop using the mailing list as a bug tracker. All issues should be put into jira and managed with that. Secondly, Didn't I ask you multiple times to stop using release snapshots and please use the SVN trunk? I don't understand why you keep ignoring me and using everything but

Re: [Freeswitch-users] Freeswitch freezes on increasing call traffic

2009-02-05 Thread paul.degt
Look like you use Fedora. I had a lot of issues with using Fedora as production or load test system, in my opinion it's more like work in progress than a production ready stable linux. If you cannot buy RHEL or SLES use Centos. shehzad p wrote: Hi Brian, As it can be seen from the system

Re: [Freeswitch-users] shoutcast skips

2009-02-05 Thread Anthony Minessale
I changed the code in tree that probably fixed it in both cases. It should be good now. On Thu, Feb 5, 2009 at 3:20 PM, Brian West br...@freeswitch.org wrote: Anyway we can look closer at the ubuntu issue also? /b On Feb 5, 2009, at 3:11 PM, e schmidbauer wrote: Hey just wanted to

Re: [Freeswitch-users] Caller ID not being passed

2009-02-05 Thread Brian West
Try application export /b On Feb 5, 2009, at 4:19 PM, Nik Middleton wrote: Try as I might, I cannot seem to get caller ID passed to the external sip gateway This GW happily processes caller id from Asterisk If tried adding param name=caller-id-in-from value=true in gw definition, and

Re: [Freeswitch-users] Caller ID not being passed

2009-02-05 Thread Brian West
Nope still doing it wrong. Try this: use export instead of set. action application=export data=effective_caller_id_number=0753960/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org

Re: [Freeswitch-users] Caller ID not being passed

2009-02-05 Thread Brian West
Nik, Ignore me... set should have worked... You're using caller-id-in-from let me look closer at this. /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote: No good, I tried action application=set data=effective_caller_id_number=0753960/ action application=export

Re: [Freeswitch-users] Caller ID not being passed

2009-02-05 Thread Brian West
Nik, While I'm looking at this can you post your full gateway and dialplan for us to see? /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote: No good, I tried action application=set data=effective_caller_id_number=0753960/ action application=export data=effective_caller_id_number/

Re: [Freeswitch-users] Caller ID not being passed

2009-02-05 Thread Nik Middleton
Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers extension name=dial_Nik_mobile condition field=destination_number expression=^(0\d+)$ action application=set

Re: [Freeswitch-users] Caller ID not being passed

2009-02-05 Thread Brian West
I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote: Dial plan is as per default setup with the addition of the following. To be honest, and I’m no SIP guru, I can’t see the caller- id being set in the sip

Re: [Freeswitch-users] Caller ID not being passed

2009-02-05 Thread Nik Middleton
Yes, I'll report back tomorrow, Regards From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 23:21 To: freeswitch-users@lists.freeswitch.org Subject:

Re: [Freeswitch-users] Transcoding G723

2009-02-05 Thread Steve Underwood
Brian West wrote: well that'll not scale far :P That transcoding card does 120 channels. A modern quad core CPU with a well implemented codec can do several hundred. A dual quad core chassis can do twice as much. Which one has a scaling problem? Steve /b On Feb 5, 2009, at 2:52 PM,

Re: [Freeswitch-users] Transcoding G723

2009-02-05 Thread Brian West
The hardware in this case... which is why I said it wouldn't scale far :P /b On Feb 5, 2009, at 5:32 PM, Steve Underwood wrote: That transcoding card does 120 channels. A modern quad core CPU with a well implemented codec can do several hundred. A dual quad core chassis can do twice as

Re: [Freeswitch-users] Transcoding G723

2009-02-05 Thread Moises Silva
At least 1 company is using it in their FS gateway for a call center of around 125 positions with their this scenario: Asterisk servers IAX G711 FS Gateway --- SIP G729 --- SIP Provider The G723 has only been tested in my laptop with an IAX connection to the FS server though. Any