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

[Satish Patel]

Why nat=yes ?

--
Sent from my iPhone

On May 2, 2011, at 7:33 AM, Tarek Sawah <tareksawah at hotmail.com> 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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