Good question. I'm not sure. I was on the forums looking for an answer to another question when I added this.
I added the text below: First post so bare with me. I've been searching off and on all day today with nothing to show for it. So now I'm posting for some help. For starters my build is sipXecs (4.6.0. 2012-09-06EDT13:09:10 ip-10-72-43-110.ec2.internal) 32bit freshly installed today (9/8/12) via iso. yum update ran earlier. 2x quad processors | 16gb RAM | plenty of hdd space. Now down to the problem at hand. Incoming calls hit a dummy "ring user" to supply the ringing to the caller (per client request) for 7 seconds. I can't find another way of getting this accomplished - open to suggestions. From there, the call is transferred to an auto attendant that acknowledges the caller and gives them some direction. If they don't chose an option they are then transferred to openACD queue and the call then drops. I watched in the openACD dashboard and the calls drop at 20 seconds - no more, no less. I've also called from an internal extension without any problems at all and no drops. I have also set up a direct DID to the queue line and called from an external phone and that too works flawlessly. This leads me away from a RE-INVITE possible issue from my ITSP. Please point me in the direction or help me in any way possible. This is the only hangup I have had with this installation so far. Thank you in advance. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Tuesday, September 11, 2012 10:34 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Calls dropping Where is the original post? On Tue, Sep 11, 2012 at 11:17 AM, Geoff Musgrave <geoff.musgr...@cacionline.net<mailto:geoff.musgr...@cacionline.net>> wrote: I have contacted my ITSP and we will be doing some troubleshooting today to see if they are causing anything or if they are seeing any strange. I'll post back with any significant results and especially if we find a resolution. I'm still open to any suggestions anyone here on the forums may have. And please let me know if you need any further details from me. thanks again. _______________________________________________ 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: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! [Description: Image removed by sender.]<http://sipxcolab2013.eventbrite.com/?discount=tony2013> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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