<div dir="ltr"><span id="result_box" class="" lang="en"><span class="">I already</span> <span>tried </span></span><span id="result_box" class="" lang="en"><span>disabling timers</span><span>,</span> <span class="">does not work.</span></span></div><div class="gmail_extra"><br><div class="gmail_quote">2016-04-08 17:19 GMT+03:00 Oleg Stolyar <span dir="ltr"><<a href="mailto:olegstolyar@gmail.com" target="_blank">olegstolyar@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Try disabling session timers in the sip profile. I think that line is commented out by default, so uncomment it.<div><br><div><param name="enable-timer" value="false"/><br></div></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On Fri, Apr 8, 2016 at 6:59 AM, Стас Тельнов <span dir="ltr"><<a href="mailto:stasan89@gmail.com" target="_blank">stasan89@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr">Hello.<br><br>When using a call or conference through sip — freeswitch with external provider there is a problem – the call is interrupted in 30 seconds. Though the sound goes all right.<br>I think that it caused by the NAT settings for freeswitch, but I don't understand how to adjust it correctly.<br>At start of freeswitch I see the following mistakes in the tracking data:<br><font size="2">2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT<br>2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5<br>2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5<br>2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5<br>2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5<br>2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5<br>2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP [general error]<br>2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP<br>2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT devices detected!</font><br><br>Despite of this mistake, conference communication between two internal users works normally. The problem arises at a call through external provider.<br><br>We have the following architecture:<br>In a cloud of Amazon EC2 there are 2 servers – opensips and freeswitch, both for NAT for external clients, but have an opportunity to work with each other directly.<br>opensips has the internal address 172.31.0.169 and external 52. *.*.177<br>freeswitch has the internal address 172.31.22.124 and external 52. *.*.198<br><br>In fact, freeswitch acts only for conferences, and is ready for use of a remote DB on opensips.<br>The auto-nat settings by default didn't work. The problem is in the external profile settings as far as I understand.<br><br>I have filled and created the following configuration:<br>vars.xml <br> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto”/><br> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198”/> <!— public freeswitch ip —><br> <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198”/> <!— public freeswitch ip —><br> <!-- External SIP Profile --><br> <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/><br> <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/><br> <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/><br> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/><br> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/tls"/><br><br>sip_profile/external.xml<br> <param name="rtp-ip" value="$${local_ip_v4}"/><br> <param name="sip-ip" value="$${local_ip_v4}"/><br><br> <param name="ext-rtp-ip" value=“52.*.*.198”/> <!— public freeswitch ip —><br> <param name="ext-sip-ip" value=“52.*.*.198”/> <!— public freeswitch ip —><br>In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server (that would be logical), but in that case conferences didn't work at all and errors below appeared:<br>[ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...<br>Also I tried to put such configuration:<br><span style="color:rgb(0,0,0)"> <param name="rtp-ip" value="auto"/><br> <param name="sip-ip" value="52.*.*.198”/></span><br>but it also hasn't helped to solve the problem.<br><br>autoload_configs/switch.conf.xml <br> <param name="rtp-start-port" value="16384"/><br> <param name="rtp-end-port" value="32768"/><br><br>"sofia status" looks as follows:<br> Name Type Data State<br>=================================================================================================<br> 172.31.22.124 alias internal ALIASED<br> external profile sip:mod_sofia@52.*.*.198:5060 RUNNING (0)<br> external profile sip:mod_sofia@52.*.*.198:5061 RUNNING (0) (TLS)<br> external::*********.com gateway sip:USER@*********.com REGED<br> internal profile sip:mod_sofia@52.*.*.198:5080 RUNNING (0)<br> internal profile sip:mod_sofia@52.*.*.198:5081 RUNNING (0) (TLS)<br>=================================================================================================<br>2 profiles 1 alias<br><br>"sofia status profile external" looks as follows:<br>=================================================================================================<br>Name external<br>Domain Name N/A<br>Auto-NAT false<br>DBName sofia_reg_external<br>Pres Hosts <br>Dialplan XML<br>Context public<br>Challenge Realm auto_to<br>RTP-IP 172.31.22.124<br>Ext-RTP-IP 52.*.*.198<br>SIP-IP 172.31.22.124<br>Ext-SIP-IP 52.*.*.198<br>URL sip:mod_sofia@52.*.*.198:5060<br>BIND-URL sip:mod_sofia@52.*.*.198:5060;maddr=172.31.22.124;transport=udp,tcp<br>TLS-URL sip:mod_sofia@52.*.*.198:5061<br>TLS-BIND-URL sips:mod_sofia@52.*.*.198:5061;maddr=172.31.22.124;transport=tls<br>HOLD-MUSIC local_stream://moh<br>OUTBOUND-PROXY N/A<br>CODECS IN PCMA<br>CODECS OUT PCMA<br>TEL-EVENT 101<br>DTMF-MODE rfc2833<br>CNG 13<br>SESSION-TO 0<br>MAX-DIALOG 0<br>NOMEDIA false<br>LATE-NEG true<br>PROXY-MEDIA false<br>ZRTP-PASSTHRU true<br>AGGRESSIVENAT false<br>CALLS-IN 0<br>FAILED-CALLS-IN 0<br>CALLS-OUT 0<br>FAILED-CALLS-OUT 0<br>REGISTRATIONS 0<br><br><br><br>What do I adjust wrong? Whether there is some opportunity, to tell freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?<br></div>
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