Tony, thank you for your response and your questions. I hope I answer them with enough detail.
My ITSP is TouchTone and they are signaling on 5080. Yes, the call will connect to the AA and will transfer to the queue and the caller will hear the MOH for a couple seconds and then the call is dropped so I have no way to confirm that audio is passing both ways because the drop occurs so quickly.. When calling directly to the queue the call works properly and audio is heard both ways, but the caller loses their options presented to them in the AA. Trunking is a SIP trunk to IP over 5080 - nothing fancy. sipXecs is behind a watchguard firewall (to an extent) but in the dmz, no alg in use all custom rules allowing full access to and from the ITSP and internal phones/softphones. sipXecs does NOT handle DNS or DHCP. SRV records are in place and when I run DNS advisor everything but the NAPTR records are working (windows DNS). The dummy ringing is a request by the client. They want their callers to hear actual "ringing" when they are calling in. I can't seem to find another way of making that sound occur rather than tying the DID to an extension and adding it to a softphone on another computer and forwarding the calls after X amount of time to the actual AA. When the DID was direct to the AA there was no "ringing" sound and they didn't like it. Thus is life. Anything else you need to know? I'm open to suggestions. Thank you! From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Tuesday, September 11, 2012 12:03 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Calls dropping - Email found in subject Explain who the ITSP is and how you have trunking setup. Typically calls from providers come to your system on port 5080. I would assume this is working because the call DOES transfer to the AA (and audio is heard, correct)? If not it is likely the call is not coming to the system on port 5080, which means the system is not properly anchoring the call. Explain your trunking and firewall. Explain if the transfer goes to the AA and audio is heard. Don't understand the use case to provide dummy ringing, so perhaps you can elaborate. On Tue, Sep 11, 2012 at 12:57 PM, Geoff Musgrave <geoff.musgr...@cacionline.net<mailto:geoff.musgr...@cacionline.net>> wrote: 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> [mailto: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! <http://sipxcolab2013.eventbrite.com/?discount=tony2013> [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 _______________________________________________ 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! [http://www.ezuce.com/image/image_gallery?uuid=61c95dd3-a26d-4363-95b1-131231e1edf0&groupId=284283&t=1340112036507%22+style=%22width:+310px;+height:+310px;]<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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