September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Weekly Conference Starting, Please Call In!
by Michael Collins 11 Sep '09

11 Sep '09

FYI, the conference is starting. Please join us! sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> 213-799-1400 -MC

3 2

Missing sofia.conf?
by Mark Sobkow 11 Sep '09

11 Sep '09

When I try to d a load mod_sofia, I get an error message indicating that Freeswitch can't find sofia.conf. There _is_ a sofia.conf.xml in the autoload directory, which I _thought_ was the main sofia configuration file. Do I need to copy it to sofia.conf? If so, where do I copy it to?

2 1

Late codec negotiation: any drawbacks?
by Carlos S. Antunes 11 Sep '09

11 Sep '09

Hello! As I have a fax machine connected to an adapter that does T.38 (Grandstream HandyTone 502), I am playing with late codec negotiation and proxy media. However, because late codec negotiation is a profile-wide affair, I would like to know if there are any potential drawbacks I should be aware of. Thanks! Carlos

1 0

Re: [Freeswitch-users] Inbound Gateway Call Not Working
by Jerry Richards 11 Sep '09

11 Sep '09

By the way, the FS DEBUG console is saying the following when an inbound call is made: Rejected by acl "domains". Falling back to Digest auth. Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Friday, September 11, 2009 10:25 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Inbound Gateway Call Not Working I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 1112223333), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username "Anonymous" and the uri "sip:4000@192.168.72.38" (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry

2 1

Inbound Gateway Call Not Working
by Jerry Richards 11 Sep '09

11 Sep '09

I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 1112223333), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username "Anonymous" and the uri "sip:4000@192.168.72.38" (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry

2 1

Call Transfer Problem
by DJB 11 Sep '09

11 Sep '09

I have a call transfer problem with Freeswitch Here is the call flow: I call from the PSTN (A party) into my Polycom phone (B-party) which is registered to FreeSwtich. The Freeswtich is setup not to route media as I have an SBC acting as a mirror proxy that will do all the NAT and media routing. The inbound call is setup fine and there is two way voice. I then blind transfer from the Polycom to my Cell phone. I see the polycom send a SIP refer to Freeswitch and it sends a 202 accepted fine and that leg between the Polycom (B party) and the A party is torn down fine like its supposed to be. The Freeswitch places the outbound call (the number the call is transferring to C-party) and that call completes. However now there is one way audio between the A party and C party . I see RTP streaming back from the egress carrier where the call was transfered to so the A party can hear the C party but the C party cannot hear the A party . When I look at the SIP traces of the original inbound call from the A-party I see a SIP re-invite from free switch to place the call on hold (contains Freeswitch RTP address to I can hear hold music) while it is transferring the call and the A-party does hear on hold music from Freeswitch while the call is being transferred. However I do not see a second re-invite from freeswitch to pass the media IP it got from the egress leg back to the original inbound leg. If my inbound gateway does not get a re-invite from Freeswitch to redirect its media to the new RTP address of of the egress carrier it will not do so hence the one way voice. How do I get the Freeswitch to re-invite the original ingress leg once it gets the SIP 183 from the egress with the new RTP info ? Free switch is sending the first SIP re-invite to my inbound gateway with new media IP (IP of itself) so the A-party can hear on hold music , but does not send a second re-invite to my inbound gateway after it receives the new RTP address from the egress carrier for the call that was transferred back out. Thank you.

2 1

XML Dial Plan vs Language Modules
by Muhammad Shahzad 11 Sep '09

11 Sep '09

Hi, I couple of my team members are working on translating a very long Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below, http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Vari… The dial plan variables are not getting initialized as expected. I was just wondering if we move this variable get and set stuff to any language module say mod_perl, will that make any difference performance wise? I mean we will be invoking a Perl interpreter for each incoming call, won't that be expensive in terms of RAM and CPU usage and thus reducing number of calls this FS deployment can handle? I have guys with programming skills in Perl, PHP, Python, Java and LUA languages. Which language do you recommend for this, again in terms of speed and performance? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

5 6

Run a command on event
by Mathieu Parent 11 Sep '09

11 Sep '09

Hi, I wanted to run an "originate" command when a MESSAGE_WAITING event is fired. Is there a simpler way than creating a daemon listening to the event socket ? Thanks Mathieu Parent

5 6

mod_xml_curl.c Oversized file detected [1056100 bytes]
by Peter P GMX 11 Sep '09

11 Sep '09

Hello, in a B2BUA scenario we have 2000 defined gateways (defined but not registered yet). When reloading mod_sofia Freeswitch complains about the XML-Curl File size > 1MB and deactivates all gateways: mod_xml_curl.c:121 Oversized file detected [1056100 bytes] Is there any way to overcome this? Currently we have 2000 gateways defined. Finally we will have about 10.000. And we will not be able to reduce the file size below 1 MB. It will become ~ 2-3 MB maybe. Best Regards Peter

4 4

uuid_exists - does it still exist?
by Phillip Jones 11 Sep '09

11 Sep '09

Hi there: The wiki states: uuid_exists Checks whether a given UUID exists. Usage: uuid_exists <uuid> However when I call via an API call I get: INVALID COMMAND! I also don't see it in MOD_COMMAND.C Has it gone? Thanks Phillip Jones.

4 7