[asterisk-users] out of the blue one way audio

[Tarek Sawah]

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_lx at hotmail.com
> To: asterisk-users at 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
> >
> >
> >
> >
> > --
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> >
>
> --
> _____________________________________________________________________
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> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
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