Thanks for the pointers! The MP-118 is pure FXO. This is a temporary solution until AT&T gets their IP Flex product installed at our site. Saga unto itself..... :)
I wiped the config back to factory, set the IP, and then tried the generated config out of sipX. As I would suspect, no go. I started comparing other posted ini files and ran across a few areas that I added in. Unfortunately the AudioCodes documentation is pretty slim so what the various options were doing are pure guesses. Where it stands now is I have inbound calls working properly!!! All calls are routed to the AA using "Endpoint Settings" | "Automatic Dialing" and setting all 8 lines to have a "Destination Phone Number" of "100". I don't think I want to port back to 5.4 at this time, but thank you for the offer. The next two issues are outgoing calls. All I get is the fast busy from my internal phones when I dial 9 and the phone number. Tried both with and without the 1 after the 9 since we do have local 10 digit dialing. Next is how do I get an inbound call to: 1) During the day route to a real user and if that user does not pick up, go to the AA 2) Off hours go directly to AA Great system! Just wish AT&T had not dropped the ball and not gotten the IP Flex up and running. Trying to get sipX up and running for Tuesday AM to replace our Cisco CM setup. Thanks again for the assistance! Andrew > -----Original Message----- > From: Todd Hodgen [mailto:[email protected]] > Sent: Sunday, January 17, 2010 5:23 PM > To: 'Josh Patten' > Cc: 'Andrew Cotter'; [email protected] > Subject: RE: [sipx-users] AudioCodes MP-118 and SipX > > Nice catch Josh, thanks. > > > > -----Original Message----- > From: Josh Patten [mailto:[email protected]] > Sent: Sunday, January 17, 2010 2:21 PM > To: Todd Hodgen > Cc: 'Andrew Cotter'; [email protected] > Subject: Re: [sipx-users] AudioCodes MP-118 and SipX > > 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/
