Since calls to VM are working, it seems the license is installed correctly. Unless your internal
calls are not using G729. A look at wireshark will quickly show what is wrong here. If it is a codec problem, you will normally see "488 Not acceptable" Sven ________________________________ From: [email protected] [mailto:[email protected]] On Behalf Of Tony Graziano Sent: 01 November 2010 21:04 To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] G.729AB - Freeswitch did you verify the codec is running and license is available using the fs_cli? On Mon, Nov 1, 2010 at 4:47 PM, James Black <[email protected]> wrote: Hi Sven, Thanks, have tried that too. No luck. It's almost like the call is not transferred/answered by voicemail ITSP, or its answered and then it hangs up. The reason that I have to use G.729AB is that it is the only Codec the ITSP uses. Kind regards James Black From: Sven Evensen [mailto:[email protected]] Sent: 01, November, 2010 2:07 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] G.729AB - Freeswitch Hi James, Check the SIP trunking config, advanced settings. The Permitted Codecs does not include G729 by default Either leave the field blank, or add G729. Sven ________________________________ From: [email protected] [mailto:[email protected]] On Behalf Of James Black Sent: 01 November 2010 11:22 To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] G.729AB - Freeswitch Hi Arman, I have subsequently installed with the http://files.freeswitch.org/g729/fsg729-153-installer The AA/Voicemail is now working for internal G729 Calls (over our vpn), however when I get calls coming in from our ITSP, as soon as the call is supposed to get answered, it is just cut. Any ideas? If someone answers the inbound call from the ITSP, it works fine, it is only when going to voicemail etc that it is cut. Kind regards James Black From: Arman Güngör [mailto:[email protected]] Sent: 29, October, 2010 1:13 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] G.729AB - Freeswitch Hi, What is the version of fsg729-xxx-installer? Arman On Fri, Oct 29, 2010 at 8:24 AM, James Black <[email protected]> wrote: Hi All, I am trying to install a licensed g729ab codec from freeswitch.org onto our sipXconfig (4.2.1-018971.dhubler 2010-08-21T04:59:18 build34) This is so that our incoming calls can terminate to voicemail / AA. In South Africa all the ITSPs use g729. I have followed instructions here: http://wiki.sipfoundry.org/display/xecsuser/G729+License and here: http://files.freeswitch.org/g729/INSTALL.txt on trying to load the new g729 module, mod_com_g729 I get the following error: freeswi...@internal> load mod_com_g729 -ERR [module load file routine returned an error] 2010-10-29 07:48:07.031712 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_com_g729.so **Trying to load an out of date module, please rebuild the module.** Are there any solutions for this? Thank you for all the wonderful support in the past! Kind regards James _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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