because they are behind a router and using private IP addresses. and the Cisco router is Nating our traffic
Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 ---------------------------------------- > From: satish...@hotmail.com > To: asterisk-users@lists.digium.com > Date: Mon, 2 May 2011 08:11:23 -0400 > Subject: Re: [asterisk-users] out of the blue one way audio > > Why nat=yes ? > > -- > Sent from my iPhone > > On May 2, 2011, at 7:33 AM, Tarek Sawah wrote: > > > > > Greetings List. > > we're facing a strange case with my system where in the middle of > > the call .. after like 7 minutes (not necessarily ) the callee is > > unable to hear the caller however the caller is able to hear the > > called party. the scenario is the following. > > > > 1- 15 computers running Windows XP SP3 joining a Windows Domain > > Controller with DHCP , DNS, ISA Internet Acceleration Server. > > 2- Internet link of 1Mbps Dedicated Leased Line. > > 3- Cisco Router > > 4- Hosted Asterisk server (Asterisk 1.4.40.1 x64 bit 8 GB ram, Intel > > (R) Xeon(R) X3210 @ 2.13GHz CPU) > > 5- additional SIP Soft phones in several locations over the world > > (Zoiper, X-Lite, Nokia Native Sip). > > 6- Packet8 Sip trunking for Inbound calls > > 7- IDT (Net2Phone) Sip Trunk for outbound calls. (two IPs) > > > > Network Profile: > > Cisco Router has a Public IP of 196.XXX.XXX.XXX and a private IP > > 192.168.100.245 > > computers have IP addresses : 192.168.100.XXX/24 > > default gateway: 192.168.100.245 > > DC: 192.168.100.2 > > DNS: 192.168.100.2 > > PROXY Server: 192.168.100.2 (Forced in Internet Explorer) > > Voip Traffic going directly from 192.168.100.245 > > Http Traffic goes to 192.168.100.2 then via another internet link > > (ADSL 8bps connection) > > > > Router is preventing any traffic other than VoIP. for example we > > tried to pass HTTP requests via the internet link .. but did not go > > through. > > > > > > Asterisk Side: > > sip.conf sample: > > [GENERAL] > > notifyringing=yes > > notifyhold=yes > > limitonpeers=yes > > tos_sip=cs3 > > tos_audio=ef > > tos_video=af41 > > alwaysauthreject=yes > > t38pt_udptl = yes > > bindport=5070 > > externip=SERVER_IP > > rtptimeout=60 > > session-timers=originate > > session-expires=600 > > session-minse=90 > > session-refresher=uas > > rtpholdtimeout=120 > > rtpkeepalive=20 > > allow=gsm > > t38pt_udptl=yes > > sendrpid=yes > > trustrpid=no > > directrtpsetup=yes > > > > [USERNAME] > > deny=0.0.0.0/0.0.0.0 > > type=friend > > secret=PASSWORD > > qualify=yes > > port=5060 > > permit=0.0.0.0/0.0.0.0 > > nat=yes > > host=dynamic > > dtmfmode=rfc2833 > > disallow=all > > allow=gsm > > context=from-callcenter > > canreinvite=no > > > > > > we have a call recording for outbound and inbound calls. > > the problem is not happening on all calls at once.. it happens on > > random > > extensions at random times and random durations however most > > noticeable durations are around 7 minutes and 20 minutes (most > > occurring) > > > > one additional situation.. the original bind_port for asterisk > > server is 5060 however after three or four hours of operating on > > that port the computers unregister and are unable to make calls at > > all .. or even register > > we changed the port to 5070 and things are working properly now. > > although this port issue is only noticeable on the above setup and > > on that facility only. other internet links are able to provide > > stable connection over 5060. > > > > any additional information can be provided. > > > > > > Tarek Sawah > > > > Information Technology Adviser > > > > Integrated Digital Systems > > > > CCNP, MCSE, RHCE, TELECOM > > > > USA: +1 386 492 9993 > > > > > > > > > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users