You mentioned you are running version 5.6 of the firmware. If you generate a configuration in sipX (which supports version 5.4) then it's not going to work properly on version 5.6. Try loading version 5.4 on your device and trying again. If you need this firmware I can send it to you off-list.
Todd Hodgen wrote: > You haven't indicated how you have programmed the MP-118, so not clear if > you manually configured it, or used a configuration from sipXecs. And, is > this a purely FXO product and not mixed fxo/fxs? > > My recommendation is to start with Sipxecs and configure the MP-118 with > their template. Once this is done, you can download the xml file by > clicking on it, and then upload it to the MP-118. Burn it and then reboot > it. This should get you a base configuration working that you can merely > tweak on the gateway. > > For your dialing, you can control that with your dial plan for that gateway. > > Personally, rather than pouring through logs, I'd either get a capture with > wireshark, and in wireshark go to Telephony/Voip Calls. It will open a > dialogue box that allows you to watch calls as they happen. You can click > on any call and do a "graph", which will allow you to see the SIP events, > and you can open an individual packet to inspect what it is doing. > > The other option is to clear all of your log file at /var/log/sipXecs. > Recreate the event, and then run /var/log/sipXecs/merged-logs. This will > create a merged.xml file. Open this file either in your os, or windows with > the SipViewer which can be downloaded from the www.sipfoundry.org website. > You can view the sip events with this viewer, and look at the individual > packet information as well. > > This should get you going down the happy path.............. > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Andrew Cotter > Sent: Sunday, January 17, 2010 8:02 AM > To: [email protected] > Subject: [sipx-users] AudioCodes MP-118 and SipX > > Good morning, > > I am having an issue with connecting an AudioCodes MP-118 (all FXO) running > firmware version 5.60A.024.003 to Sipx 4.0.4-017289 with 8 copper lines. I > have a Polycom 550 and Cisco 7960 setup for testing. > > Problem #1 > I can dial in from the outside and get the AA. From there I try and dial an > extension or dial by and the attendant says "please hold why I transfer your > call" then a few seconds later the line is disconnected. > > I am new the sipx world, but as I understand it the problem might be with > REFER? > > > Problem #2 > Having either of the phones dial out. I have added the gateway and the > built in local and long distance dial plans. At one point it would dial out > but was send the wrong combo of digits since I kept getting the "please dial > 1 and the area code" or some other message that the number was incorrect > depending on the configuration at the time. > > Not sure how to grab a good look at the logs or even what I should be > looking for. Any help would be appreciated. > > Thanks! > > Andrew > > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
