The keep-alive may actually be important for your local firewall to keep connections open...
Mike On Mon, Sep 6, 2010 at 6:37 PM, dan <d...@dm3.us> wrote: > On the Internet Calling->NAT Traversal page I have checked the box next > to "Server is behind NAT." > > On the Gateways -> ISTP Account page SIP Keepalive is set to "None" > which is supposed to default to CR-LF and RTP keepalive is set to > "replay last packet." > > Flowroute's account settings page doesn't offer any options regarding > NAT. I'll have to ask them if they can disable any NAT Traversal > settings. > > > On Mon 06.Sep.10 17:28, Tony Graziano wrote: > >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/ > > > > -- > 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/ > -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpic...@gmail.com blog: http://www.sipxecs.info call: sip:mpic...@sipxecs.info <sip%3ampic...@sipxecs.info>
_______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
