Group: I have a customer that is running the following Asterisk CVS-HEAD dated 2005-08-18 WhitBox Linux respin 2 mysql Ver 11.18 Distrib 3.23.58 Cisco 7960G We are using the real-time drivers for sip and everything is working great. They have a few employees that use the phones from home on a RR or DSL line. The problem is if they make a call everything works great they hang up and are able to get inbound calls. If they do not make a call for 5 or 10 mins they are unable to get inbound calls. If they dial out again its all working for another 5 or 10 mins. This does not happen to all remote people just a few. Anyone have any ideas what the heck is going on with this? Thanks for your time.
Hall, Eric M. wrote:> Asterisk CVS-HEAD dated 2005-08-18 > WhitBox Linux respin 2 > mysql Ver 11.18 Distrib 3.23.58 > Cisco 7960G > > We are using the real-time drivers for sip and everything is working > great. > They have a few employees that use the phones from home on a RR or DSL > line. > The problem is if they make a call everything works great they hang up > and are able to get inbound calls. If they do not make a call for 5 or > 10 mins they are unable to get inbound calls. If they dial out again its > all working for another 5 or 10 mins. This does not happen to all remote > people just a few.Using Realtime SIP peers does not allow for "NAT Keepalive" packets to be sent, so the firewall/NAT devices that those phones are connected to are closing the SIP port hole after an expiration timeout. To fix this, you'll need to upgrade to newer Asterisk (you really should be running 1.2) and use 'realtime caching' for your SIP peers.
I'm using realtime caching. Here is my sip.conf file [general] callerid=unavailable context=default allowguest=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no rtcachefriends=yes allow=ulaw allow=g729 All other information about the sip clint is keep in the db Thanks again! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, February 14, 2006 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nat, SIP, Realtime problem Hall, Eric M. wrote:> Asterisk CVS-HEAD dated 2005-08-18 > WhitBox Linux respin 2 > mysql Ver 11.18 Distrib 3.23.58 > Cisco 7960G > > We are using the real-time drivers for sip and everything is working > great. > They have a few employees that use the phones from home on a RR or DSL> line. > The problem is if they make a call everything works great they hang up> and are able to get inbound calls. If they do not make a call for 5 or> 10 mins they are unable to get inbound calls. If they dial out again > its all working for another 5 or 10 mins. This does not happen to all > remote people just a few.Using Realtime SIP peers does not allow for "NAT Keepalive" packets to be sent, so the firewall/NAT devices that those phones are connected to are closing the SIP port hole after an expiration timeout. To fix this, you'll need to upgrade to newer Asterisk (you really should be running 1.2) and use 'realtime caching' for your SIP peers. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Just wanted to also say this does not happen to all users behind a NAT box on RR or DSL line just a few. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, February 14, 2006 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nat, SIP, Realtime problem Hall, Eric M. wrote:> Asterisk CVS-HEAD dated 2005-08-18 > WhitBox Linux respin 2 > mysql Ver 11.18 Distrib 3.23.58 > Cisco 7960G > > We are using the real-time drivers for sip and everything is working > great. > They have a few employees that use the phones from home on a RR or DSL> line. > The problem is if they make a call everything works great they hang up> and are able to get inbound calls. If they do not make a call for 5 or> 10 mins they are unable to get inbound calls. If they dial out again > its all working for another 5 or 10 mins. This does not happen to all > remote people just a few.Using Realtime SIP peers does not allow for "NAT Keepalive" packets to be sent, so the firewall/NAT devices that those phones are connected to are closing the SIP port hole after an expiration timeout. To fix this, you'll need to upgrade to newer Asterisk (you really should be running 1.2) and use 'realtime caching' for your SIP peers. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users