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: 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: 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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