Chris,
After reading this through a bit more... how about trying the following. It looks like you really don't need a hunt group. Why don't you trash the 601 hunt group, create a user named 601, disable voicemail, set forwarding to forward to extension 8218 after 30 seconds. Put extension 601 on all 4 of those phones. You can also then give that extension a different ring tone vs. their normal extension ringing. Mike From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen Sent: Saturday, January 02, 2010 8:46 PM To: 'Chris Rawlings'; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Fwd: Inbound Call not hanging up Why do you have the Alias answered by a Phantom Ext, and then forward to an Autoattendant? You can place that Alias number in the Auto-Attendant so that it goes directly to the Auto-Attendant and accomplish the same thing. Doubt if this is related to your issue, but seems like an extra unnecessary step, unless I'm missing some other requirement. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Chris Rawlings Sent: Saturday, January 02, 2010 4:35 PM To: sipx-users@list.sipfoundry.org Subject: [sipx-users] Fwd: Inbound Call not hanging up Who is your ITSP or gateway type? i am using Bandwidth.com <http://Bandwidth.com/> with the SipX Gateway/SBC What model phone (include firmware and bootrom versions)? Polycom IP450 3.1.3 REV C firmware What lines you are configuring separately for hunt groups? its not only my Hunt Groups that are Fubar and they are only fubar if they include MoH The SBC is setup with a blank extension to drop calls to The SBC is setup with MoH checked Incomming call comes into a Phantom Ext that has an alias set to the DID that is comming in That call is then call fowarded to an Auto Attendant The AA has options on it.. option 1 being the most used one... when option 1 is pressed that is fowarded to ext 601.. Ext 601 is a Hunt Group The hunt group does not allow call forwarding and i have the option set if no one answers it is fowarded to ~~vm~...@voip.royalshockey.com The users in the hunt group are setup like so 216 ring for 30 seconds 218 ring "at same time" for 30 seconds 219 ring "at same time" for 30 seconds 220 ring "at same time" for 30 seconds after 30 seconds it gets forwarded to ~~vm~...@voip.royalshockey.com where VM's are left i NEVER have 1 way audio issues I DO have the SIP signaling set to 5080 Issue is that when someone calls in and hangs up before the call is picked up the phones in the office keep ringing. I believe if this were to be fixed it would probably fix my issues with Queues which is another discussion. On Jan 2, 2010, at 7:08 PM, Tony Graziano wrote: I think the fiewall is not the issue. It's more likely an issue with call forking and how you are configuring your inbound calls. Something is systemically wrong within sipXecs as it relates to the way your have configured hunt groups and queues. Another firewall and config will not assist you in resolving this. Identify the following: Who is your ITSP or gateway type? (answers should be an audiocodes modell xxx with POTS/PRI or ITSP xyz coming through sipxbridge) What model phone (include firmware and bootrom versions)? What lines you are configuring separately for hunt groups? (explain ONE hunt group you are having a problem with and describe EXACTLY how it is configured) On Sat, Jan 2, 2010 at 6:33 PM, Chris Rawlings <cm.rawli...@gmail.com> wrote: this is driving me nuts... this does this with every phone system i have installed using SipX. i can not be the only person that is having this issue... just setup Vyatta.. port forwarded everything... setup masquerading... same issue. Basically when i call into the system.. and hang up before the call is answered the call doesn't drop and the phones keep ringing. Doesn't matter if its a Queue or Hunt Group. i don't know anymore On Jan 2, 2010, at 6:08 PM, Tony Graziano wrote: The link I've already sent you has a downloadable configuartion file. From there you can peruse it with the pfsense gui. On Sat, Jan 2, 2010 at 6:05 PM, Chris Rawlings <cm.rawli...@gmail.com> wrote: This is the other thread that i am reffering to from my last post please if anyone has any ideas i would be all up for them... basically i KNOW how to remove MoH from polycom phones that is very easy. i do know how to setup NAT properly i believe its the device i am using to NAT with that is doing "Port Randomization".... also i do have SIP ALG turned off. i am currently using Untangle for firewalling.. if someone could tell me how to setup pfSense not with a link but what options need to be turned on or off. also how to setup the NAT.. should it be 1:1 ? also as of now i have vyatta up and running but am testing things out right now. On Fri, Dec 25, 2009 at 7:03 PM, Tony Graziano <tgrazi...@myitdepartment.net> wrote: I might be repeating myself... Polycom firmware 3.1.3, pfsense 1.2.3 and MOH disabled on the phones but not on sipxbridge. Always works. Here's the pfsense howto... http://blog.myitdepartment.net/?p=37 You will also find a fairly easy to follow for bandwidth.com <http://bandwidth.com/> and siptrunking. Many, many have followed these without issue. ============================ 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: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> Sent: Fri Dec 25 18:54:03 2009 Subject: Re: [sipx-users] Inbound Call not hanging up can you link that how-to for pfSense... i'll give something else another try. at this point tho... the only time this happens is if MoH is being played... if i dial into something that does not have MoH being played it hangs up properly. also it will hang up at the end of the MoH recording right before the recording starts to play again it sees there is no connection from the ITSP and it finally hangs up why would the server not hang up tho if it see's the BYE from the ITSP On Fri, Dec 25, 2009 at 6:12 PM, Tony Graziano <tgrazi...@myitdepartment.net > wrote: > If you follow the how-to I wrote to connect to bandwidth.com <http://bandwidth.com/> using a > pfsense > firewall it will work. If it does not, a trace would be useful. It's more > likely an ISP issue but not something that can be proved without a trace. > > > ============================ > 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: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> > Sent: Fri Dec 25 13:37:22 2009 > Subject: Re: [sipx-users] Inbound Call not hanging up > > Standard comcast cable modem in bridged / routed mode > > i don't know if they use transient networks on the Cable side to set this > device up in route mode or if they use a bridged device mode.. but the > comcast modem is setup to do 0 firewalling / NAT. > > i logged into the device and made sure of this > > On Fri, Dec 25, 2009 at 1:25 PM, Scott Barr <stuckb...@gmail.com> wrote: > > > We are using sipX 4.04 with bandwidth.com <http://bandwidth.com/> , as well, after following > Tony's > > most > > helpful guides. We do not experience these issues you have described and > > we > > use > > several different Polycom phone models. > > > > What device is between your firewall and ITSP and is it configured to > > NAT > > as > > well? > > > > -Scott > > > > > > > > > > _______________________________________________ > > sipx-users mailing list sipx-users@list.sipfoundry.org > > List Archive: http://list.sipfoundry.org/archive/sipx-users > > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > > sipXecs IP PBX -- http://www.sipfoundry.org/ > > > > > > -- > Thank You, > Chris Rawlings > IT Consultant > > phone - 610.741.3324 > > VCP RHCE RHCT MCSE > -- Thank You, Chris Rawlings IT Consultant phone - 610.741.3324 VCP RHCE RHCT MCSE -- Thank You, Chris Rawlings IT Consultant phone - 610.741.3324 VCP RHCE RHCT MCSE _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ -- ====================== 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/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ -- ====================== 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/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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