you need a siptrace in this instance to see whats up. On Mon, Nov 1, 2010 at 5:20 PM, Sven Evensen <[email protected]>wrote:
> 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 > > > > _______________________________________________ > 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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