I'm partial to patton gateway, but if you already have the mediant it should be fine for this.
You can put your remote users in a group with no call permissions except internal and remove permissions for internal voicemail. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: sipx-users-boun...@list.sipfoundry.org <sipx-users-boun...@list.sipfoundry.org> To: Discussion list for users of sipXecs software <sipx-users@list.sipfoundry.org> Sent: Sat Oct 23 18:19:11 2010 Subject: Re: [sipx-users] inbound route Thanks Tony it worked by setting facility_feature_service=0 and facility_coding_value=0 in Dialogic... I need your recommendation on this if you dont mind..by the way I'm not an expert in VoIP :) I have to implement a TDM to VoIP network.. For the longest time, I have been using Asterisk and Digium and then I got my hand on an Audiocodes gateway.. So I have decided to have some remote users on this solution. Well the remote user needs to register with a SIP registrar which I have decided to use SIPx. both Sipx and users are behind two different NATs which based on what I read on Wiki this shouldn't be a problem. I also read that Sipx handles calls better than Asterisk and quality of the calls should be better in Sipx.. that's one of the reason that I want to switch from Asterisk to Sipx... Users will be receiving calls from my PBX on their IP phones/softphone only(usually long calls, 4 to 5 hours).. they cannot make any phone calls, no voice mail etc needed...PBX is patched using T1 cross over cable into Audiocodes/digium Do you think using Audiocodes and Sipx is a good solution here? I just need your thought on this, or anyone else for that matter Thanks ________________________________________ From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano [tgrazi...@myitdepartment.net] Sent: Saturday, October 23, 2010 12:16 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] inbound route I'm not a heavy AudioCodes fan, so I won't be able top provide much in the way of guidance. I don't think you need to setup the Audiocodes in sipx at all, since your use is different. What you need to ensure is making sure the Audiocodes sends the call to sipx as its destination route (tel>ip i think). For this you can simply use the IP address of sipx in the audiocodes. You should also add the internal ip address of sipx as a domain alias in sipxconfig (system>domain>alias). You need to know what the DIALOGIX is sending as digits (2,3,4,5,6,7,8,9,10,11). Example: If it is sending 1234567, and you want 1234567 to go to a user in sipx (and there is no user 1234567) specified as user 200, then add 1234567 as an alais in the user 200. You should put the sipx proxy in debug mode and tail it with: tail -f /var/log/sipxpbx/sipXproxy.log to see if the calls are getting to sipx, and view the log in the mediant to see what information it is catching about the call. On Sat, Oct 23, 2010 at 11:34 AM, David Sharafy <da...@ccds.ca<mailto:da...@ccds.ca>> wrote: Can someone please help me with this? ________________________________________ From: sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org> [sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org>] On Behalf Of David Sharafy [da...@ccds.ca<mailto:da...@ccds.ca>] Sent: Friday, October 22, 2010 1:03 PM To: sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> Subject: [sipx-users] inbound route Hi, I have an Audiocodes M 1000, firmware version 5.2 .. I have been using Asterisk before and new to Sipx.. Audiocodes has a T1 that is patched to an internal PBX (using a T1 cross over cable), the pbx is Dialogic based. I'm sending a call from Dialogic to Audiocodes and then converting it to SIP call and sending it to Asterisk (SIP user). I tried to replicate the exact setup using SipX, but the call fails and my guess is that I'm missing an inbound route or possibly something else. Basically the user registers with SipX (i.e user 201), I then send 201 from my PBX to Audiocodes and Audiocodes will send it to SIPx, at this point the phone should ring which now it doesn't. I checked "Call Detail Records" and I see that the call is reaching Sipx but the result says FAILED. I tried to look into sipx logs but can't find it anywhere, I changed logging to debug but I'm not sure which file contains the inbound calls... I also added the audiocodes to sipx as Gateway and downloaded ini and uploaded to audiocodes which works ok but still can't send the call. for this setup, I only need the ability to send a call from Audiocodes to SIPx, no need to send any calls from SIPx to Audiocodes. Thanks for your help. _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net<mailto:tgrazi...@voice.myitdepartment.net> Fax: 434.326.5325 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> Helpdesk Contract Customers: http://support.myitdepartment.net _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
