Can you confirm you system is behind nat, and your settings are as I described? ============================ 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: Tony Graziano <tgrazi...@myitdepartment.net> To: Discussion list for users of sipXecs software <sipx-users@list.sipfoundry.org> Sent: Mon Sep 06 17:08:36 2010 Subject: Re: [sipx-users] calls drop after 20 minutes With flowroute, your sip-keepalive should be CR-LF and the RTP keepalive should be "replay last packet". You can safely ignore their statement about NAT's networks, because sipxbridge takes care of that for you. In ANY installation where sipx is behind nat with a properly configured firewall YOU_SHOULD_NOT have any setting at the ITSP that activates ANY_KIND_OF_NAT_HELPER mechanism. For instance, with voip.ms I disable NAT for the account, MOH and so forth. I don;t want them to try to figure NAY of that out for me, because sipxbridge is well suited to do that for me. I don't know how to interpret the statement from flowroute, because it is confusing, as you DONT WANT them to try to traverse NAT for you in a properly installed and configured system with sipxbridge behind NAT. Verify the above settings. With flowroute, your sip-keepalive should be CR-LF and the RTP keepalive should be "replay last packet" in your gateway configuration. DISABLE ANY NAT HELPER "stuff" that flowroute allows you to, there is typically only ONE setting. If after this you have issues, ask them to capture a call that fails and tell you "why", because it would appear to be a call timer issue (on their end). On Mon, Sep 6, 2010 at 4:49 PM, Dan McDaniel <d...@dm3.us> wrote: > I asked Flowroute what was their preferred method for SIP keepalive. > Apparently, they don't need it at all. Here's the reply: > > The SIP keepalive is used by VoIP providers that cannot transverse NATed > networks, however, Flowroute's network supports the use of NATed > systems as we are able to transverse the network so long as you have not > modified your system to make it appear as if it is not NATed. > > Also, I only have two choices for SIP keepalive on the gateway->ISTP > Account page: None and Empty SIP message. The description below the box > says, > > Defines the message to use for SIP keepalive. If nothing is specified, > CR-LF (empty SIP message) is used. > > This makes it sound like "None" defaults to empty SIP message, or you > can select "empty SIP message." What's the difference? > > > On Mon 06.Sep.10 09:49, dan wrote: > >Thanks, Tony. I'll check with them. > > > >On Mon 06.Sep.10 12:07, Tony Graziano wrote: > >>You need to ask them what their preferred SIP KEEPALIVE method is, and > >>change the gateway accordingly. > >> > >>Alternatively you can change the SIP KEEPALIVE method to a different > value > >>and wait 21 minutes to see if the calls till works. If you have an > account > >>with them, they should be able to answer this easily. > >> > >>On Mon, Sep 6, 2010 at 12:04 PM, dan <d...@dm3.us> wrote: > >> > >>> > >>> I've been having a problem with calls dropping. The ISTP (Flowroute) > >>> sends a BYE after 20 minutes and the call ends. Is this more likely to > >>> be a configuration problem on my end or a problem with the ISTP? Just > >>> looking for pointers on where to start troubleshooting. > >>> > >>> I'm running 4.2.1 (4.2.1-018971.dhubler 2010-08-21T04:59:18 build34). > >>> > >>> -- > >>> Dan McDaniel > >>> d...@dm3.us > >>> Key fingerprint = CAEC B8D9 3701 86CF D3B2 1E99 D8BB F217 455C AD36 > >>> _______________________________________________ > >>> 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: 434.984.8431 > >> > >>Email: tgrazi...@myitdepartment.net > >> > >>LAN/Telephony/Security and Control Systems Helpdesk: > >>Telephone: 434.984.8426 > >>sip: helpd...@voice.myitdepartment.net > >>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/ > > > > > >-- > >Dan McDaniel > >d...@dm3.us > >Key fingerprint = CAEC B8D9 3701 86CF D3B2 1E99 D8BB F217 455C AD36 > >_______________________________________________ > >sipx-users mailing list > >sipx-users@list.sipfoundry.org > >List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > Dan McDaniel > d...@dm3.us > Key fingerprint = CAEC B8D9 3701 86CF D3B2 1E99 D8BB F217 455C AD36 > _______________________________________________ > 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: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net 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/
