February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

monitoring events in Python
by Adam Wilt 14 Feb '09

14 Feb '09

I'm trying to use custom events for a conference call in a Python script. I set-up the events in the conference.conf.xml file, and I send "bgapi event plain CUSTOM conference::maintenance" to enable them. But I don't know how to look for these events in my script. Does anybody have some example code, or maybe just point me in the right direction? Thanks.

3 2

No-media problem with opensips-freeswitch setup
by Woody Dickson 14 Feb '09

14 Feb '09

Hi, I tried to configure opensips as sip proxy and sip registrars and freeswitch as B2BUA. Everything works until I start to connect sip clients that are behind ADSL. Both freeswitch and opensips are on public IP and I am using external profile as well. Does anyone have experience in setting up opensips and freeswitch together and can share the configuration with me? Thank you very much in advance for any help. Regards, Woody

2 2

Re: [Freeswitch-users] Transcoding G723
by Mike Tkachuk 14 Feb '09

14 Feb '09

Hello Freeswitch-users, Check this one: http://freehg.org/u/deepwalker/fs_g729/ G.729 is not G.723 but may be interesting. IPP have also g.723 implementation, not too hard to port. That code is working OK for me on development servers. -- Mike

1 0

INFO: Some new content on main page
by Michael Collins 13 Feb '09

13 Feb '09

FYI, There are a few new items on the main page: www.freeswitch.org, just in case you haven't been there lately. :) -MC

1 0

speex build issues in svn trunk.
by Michael Jerris 13 Feb '09

13 Feb '09

I updated the version of the speex library we use in tree last night and it may cause some build issues for those with current working copies. To fix this issue you can type "make speex-reconf" MIke

2 1

Codec negotiation questions
by ivdreg ivdreg 13 Feb '09

13 Feb '09

Hi all, Can I ask 2 questions about codec negotiation: 1. Is it possible Freeswitch to work negotiate codecs between two phones as it is described below. INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec preference according absolute_codec_string but exclude all codecs not offered by A ----> INVITE to B with rewrited SDP. example: from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM ----> to B SDP: PCMU,PCMA 2. Can I get codec list in INVITE with mod_perl for example or via xml_curl without processing SDP variable (switch_r_sdp). It will be useful to be in format that absolute_codec_string variable takes. Thanks

2 7

Setting outbound callerid using js
by Nik Middleton 13 Feb '09

13 Feb '09

Hi Guys I'm trying to set the outbound caller-id in js. The params seem to be acceptable, except I'm getting the default +000000000 caller-ID sent. Should the below work with js? session.originate(session,'{accountcode=54321,ignore_early_media=true,or igination_caller_id_number=07630600000,originate_timeout=25}sofia/gatewa y/mygw/01XXXXXXXXXXX'); (this works using lua BTW) regards

4 8

Ruby framework for event socket
by Jan Kubr 13 Feb '09

13 Feb '09

Hi all, I've created a simple framework in Ruby that you can use to talk to Freeswitch via even socket outbound. It won't suite your needs perfectly if you are doing anything non-trivial, but it might be a nice starting point. Check it out at http://github.com/jankubr/freec Cheers, Jan Kubr

2 1

Question: SIP BYE authentication
by Helmut Kuper 13 Feb '09

13 Feb '09

Hello, my FS is connected to my SIP-DDI softswitch, which requires all SIP requests sent by a registered SIP account to be authenticated. I found that when FS sends a BYE FreeSWITCH ignores the authentication challenge (SIP/2.0 407) received from proxy and simply terminates the session. Is there a way to configure FS in that way that it react on auth challenges for BYEs ? regards Helmut

4 6

xml_cdr call flow
by Luis F Urrea 13 Feb '09

13 Feb '09

Hi all, We are writing a xml_cdr parser to load CDRs in SQLite. We are interested in logging times for both A leg and B leg so that transfers are reported as individual calls with accurate timing. eg Inboud call to AA lasted 14 seconds then call to operator 20s and then call to actual extension 5min As of now we are using the <callflow> tag with the "number" attribute to find out who did the A leg talk to, then we open the B leg files and get the times from each jump from the <times> tag within the <callflow> tag on the B leg file. Is this right or maybe someone could suggest a better way to do it. TIA Luis

3 6