"whatever" answers the call, if/when configured properly is going to ACK it. Assuming the ITSP is competent they will see the call as answered.
I simply don't know enough about your environment, but I would suggest simply having the calls route to the AA (it shows answered this way), then let it transfer to the hunt group or openacd queue. you probably need to ask questions about the best way to handle openacd when there is no agent logged in. It would be important to make sure the reinvite is being honored by the itsp by testing it with a logged in agent or to an AA and transferred to a ua that is registered. you might also double-check the sonicwall config. I wrote a little 3 step thing a year ago, because it kept coming up: http://wiki.sipfoundry.org/display/sipXecs/Sonicwall On Tue, Sep 11, 2012 at 6:47 PM, Geoff Musgrave < geoff.musgr...@cacionline.net> wrote: > I eliminated the dummy ring piece and still the same result. I was able > to verify that I’m getting 2 way audio before the call drops though. My > ITSP is being difficult, but I think it’s more because I wasn’t available > to respond to their response soon enough to further explain the situation > and clear up the obvious confusion they were having. **** > > ** ** > > I’ll see what I come up with from here but if you have any further > suggestions that I can try I’m open to them. They were getting 183 when > dialing directly to the queue and telling me that was an abandoned call > (another issue I have to deal with – suggestions?) because it was never > “answered” so they never saw a 200. Sorry, there wasn’t an agent logged in > so that’s my assumption as to the 183 but I would have thought that when > the MOH kicked in on the queue sipXecs would have sent the 200 back – I > don’t know. They determined the call through the AA was ok because they > were transferred and all the SIP messages came back properly. So now I > continue the battle. **** > > ** ** > > As for godaddy – I’m not a fan at all. I don’t even use them for a > registrar anymore (personally) but it’s what was in place here. I need to > take the 10 minutes and find a reliable host to move our DNS to or spend > the time to bring it in house. **** > > ** ** > > Again, thanks for all your help.**** > > ** ** > > *From:* sipx-users-boun...@list.sipfoundry.org [mailto: > sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano > *Sent:* Tuesday, September 11, 2012 12:47 PM > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] Calls dropping - Email found in subject - > Email found in subject - Email found in subject**** > > ** ** > > Ugh. Go daddy. i only use them for a registrar but cant even stomach their > DNS and moved everything. Good to know I missed that headache yesterday too! > **** > > On Tue, Sep 11, 2012 at 1:44 PM, Geoff Musgrave < > geoff.musgr...@cacionline.net> wrote:**** > > Thanks Tony. I was thinking that yesterday when I requested the > troubleshooting with the ITSP but I had other godaddy related fires to put > out yesterday. We will be doing some captures and go from there this > afternoon and I’ll report back. I’ll also try going direct to the AA and > we’ll see how that goes.**** > > **** > > Thanks again.**** > > **** > > *From:* sipx-users-boun...@list.sipfoundry.org [mailto: > sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano > *Sent:* Tuesday, September 11, 2012 12:35 PM > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] Calls dropping - Email found in subject - > Email found in subject**** > > **** > > It is more likely the ACK is not being OK'd by the ITSP. I would try to > simplify the call inbound to see if this applies to all calls or not.**** > > **** > > In other words, send the call to the AA (dont ring) and then see if the > call will stay up for longer than 20 seconds.**** > > **** > > Hint: If the call answers and audio flows between both ends for 20-30 > seconds but ALWAYS drops, it typically means the call is not being ACK'ed > by the ITSP and they are sending a BYE. A call trace or pcap will also bear > this out.**** > > **** > > On Tue, Sep 11, 2012 at 1:21 PM, Geoff Musgrave < > geoff.musgr...@cacionline.net> wrote:**** > > 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> 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] *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> 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 > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > > > **** > > **** > > -- > ~~~~~~~~~~~~~~~~~~ > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: 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>**** > > [image: Description: Description: 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**** > > **** > > Helpdesk Customers: http://myhelp.myitdepartment.net**** > > Blog: http://blog.myitdepartment.net**** > > > _______________________________________________ > sipx-users mailing list > 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 > 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! > *Error! Filename not > specified.*<http://sipxcolab2013.eventbrite.com/?discount=tony2013> > **** > > **** > > **** > > LAN/Telephony/Security and Control Systems Helpdesk:**** > > Telephone: 434.984.8426**** > > sip: helpd...@voice.myitdepartment.net**** > > **** > > Helpdesk Customers: http://myhelp.myitdepartment.net**** > > Blog: http://blog.myitdepartment.net**** > > > _______________________________________________ > sipx-users mailing list > 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 > 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! > [image: Description: 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**** > > **** > > Helpdesk Customers: http://myhelp.myitdepartment.net**** > > Blog: http://blog.myitdepartment.net**** > > > _______________________________________________ > sipx-users mailing list > 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 > 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! > [image: 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**** > > ** ** > > Helpdesk Customers: http://myhelp.myitdepartment.net**** > > Blog: http://blog.myitdepartment.net**** > > _______________________________________________ > sipx-users mailing list > 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 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> -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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