# Asterisk 1.6 reinvite **URL:** \ **Category:** Asterisk Support\ **Created:** [January 22, 2011, 2:02pm UTC](https://community.asterisk.org/t/asterisk-1-6-reinvite/33921 "2011-01-22T14:02:55Z")\ **Posts on this page:** 2\ **Page:** 1
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [January 22, 2011, 2:02pm UTC](https://community.asterisk.org/t/asterisk-1-6-reinvite/33921/1 "2011-01-22T14:02:55Z")
HI All. After 15 minutes of conversation asterisk sends a signal “reinvite” and stop responding. The conversation is interrupted. Please help to solve the problem … Version: asterisk16-configs-1.6.2.12-1\_centos5 asterisk16-core-1.6.2.12-1\_centos5 asterisk16-1.6.2.12-1\_centos5 asterisk16-doc-1.6.2.12-1\_centos5 asterisk16-voicemail-1.6.2.12-1\_centos5 asterisk16-dahdi-1.6.2.12-1\_centos5 asterisk-sounds-core-en-gsm-1.4.20-1\_centos5 sip.conf [general] context = default; Default context for incoming calls canreinvite = no directmedia = no allowsubscribe = no transport = udp allowguest = yes; Allow or reject guest calls (default is yes) bindport = 5060; UDP Port to bind to (SIP standard port for unencrypted UDP udpbindaddr = 0.0.0.0 ; bindaddr = 0.0.0.0; IP address to bind UDP listen socket to (0.0.0.0 binds to all) bindaddr = 217.23.132.83; You can specify port here too, like 123.123.123.123:5080 ; tcpenable = no; Enable server for incoming TCP connections (default is no) ; tcpbindaddr = 0.0.0.0 udpenable = yes srvlookup = yes; Enable DNS SRV lookups on outbound calls useragent = [prostoy.ru](http://prostoy.ru); Allows you to change the user agent string videosupport = yes; Turn on support for SIP video. You need to turn this on rtcachefriends = yes rtupdate = yes rtautoclear = yes disallow = all allow = g729 THX ALL! ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [January 24, 2011, 11:53am UTC](https://community.asterisk.org/t/asterisk-1-6-reinvite/33921/2 "2011-01-24T11:53:28Z")
This is typically the result of a problem in the peer’s handling of session timers. An actual SIP trace would probably make this clear.