the session interval timer is set for 30 minutes by default (1800 seconds).
I posted vigorously about bandwidth.com and their problems with dtmf and then not honoring ptime of 20ms and playing media services at 30ms over the two past shipping versions of sipx. I actually think all of this is due to interop issues with freeswitch (media server) and bandwidth.com rather than the trunk settings. I am going to assume because of this past research that the issue is that bandwidth.com is killing the connection because they are not getting the ack back from the media server instead of from the trunk. We used to like bandwidth.com at my place a lot, but we have since moved every customer off them for these and other reasons (some of them long outages). The rest were actually due to media server issues and not general trunking problems though. On Fri, Apr 6, 2012 at 12:49 PM, Michael Picher <[email protected]> wrote: > try playing with keepalive settings on the trunk... > > also, may be a firewall issue, not even related to SIP... > > but start with keepalive... > > On Fri, Apr 6, 2012 at 12:32 PM, Jarrod Cuzens <[email protected]> wrote: > >> We are having an issue where Bandwidth.com drops our conference call >> after about 15 minutes. Attached is the sipxtrace. I know a lot of people >> use Bandwidth.com so I'm wondering if someone can provide insight. >> >> Basically, the call goes along fine and then a REINVITE occurs and >> Bandwidth.com returns a 481. >> >> INVITE sip:[email protected]:5060 SIP/2.0^M >> Via: SIP/2.0/UDP 216.55.27.54:5080 >> ;branch=z9hG4bK004f2e385f3b8153396a7e0d3e17526f313833^M >> CSeq: 2 INVITE^M >> From: "sipxbridge" <sip:[email protected] >> >;tag=7663711578417146310^M >> To: <sip:[email protected];user=phone>;tag=gK0db35f58^M >> Call-ID: [email protected]^M >> Content-Disposition: session;handling=required^M >> Max-Forwards: 70^M >> Route: <sip:216.82.224.202:5060;lr;ftag=7663711578417146310>^M >> User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)^M >> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS^M >> Accept: application/sdp^M >> Content-Type: application/sdp^M >> Contact: <sip:216.55.27.54:5080>^M >> Session-Expires: 1800;refresher=uac^M >> Min-SE: 90^M >> Content-Length: 198^M >> ^M >> v=0^M >> o=sipxbridge 7423237425625656682 1 IN IP4 216.55.27.66^M >> s=SIP Call^M >> c=IN IP4 216.55.27.66^M >> t=0 0^M >> m=audio 30280 RTP/AVP 0 8^M >> a=sendrecv^M >> a=rtpmap:0 PCMU/8000^M >> a=ptime:20^M >> a=rtpmap:8 PCMA/8000^ >> >> SIP/2.0 100 Giving a try^M >> Via: SIP/2.0/UDP 216.55.27.54:5080 >> ;branch=z9hG4bK004f2e385f3b8153396a7e0d3e17526f313833;received=216.55.27.66^M >> CSeq: 2 INVITE^M >> From: "sipxbridge" <sip:[email protected] >> >;tag=7663711578417146310^M >> To: <sip:[email protected];user=phone>;tag=gK0db35f58^M >> Call-ID: [email protected]^M >> Server: Bandwidth.com TRM (bw7.gold.13)^M >> Content-Length: 0 >> >> SIP/2.0 481 Call Leg/Transaction Does Not Exist^M >> Via: SIP/2.0/UDP 216.55.27.54:5080 >> ;received=216.55.27.66;branch=z9hG4bK004f2e385f3b8153396a7e0d3e17526f313833^M >> From: "sipxbridge" <sip:[email protected] >> >;tag=7663711578417146310^M >> To: <sip:[email protected];user=phone>;tag=gK0db35f58^M >> Call-ID: [email protected]^M >> CSeq: 2 INVITE^M >> Content-Length: 0 >> >> Thanks, >> Jarrod >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > Michael Picher, Director of Technical Services > eZuce, Inc. > > 300 Brickstone Square**** > > Suite 201**** > > Andover, MA. 01810 > O.978-296-1005 X2015 > M.207-956-0262 > @mpicher <http://twitter.com/mpicher> > www.ezuce.com > > > ------------------------------------------------------------------------------------------------------------ > There are 10 kinds of people in the world, those who understand binary and > those who don't. > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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