4 digit dialing

[phoneproblems]

TIA
Problem-cant hear audio from site A or Site B
loginto SS- packet loss 100%
i can dial 4 digits and i hear ring back just when person picks up no audio.

Company A- IPO Server edition
h.323 line
GW> xxx.xxx.xxx.100
L3
SS:h450
allow direct media- Uncheck

Short Code
4xxx
dial
.
LG-3

Company B- IPO Server edition
h.323 line
GW. xxx.xxx.xxx.169
L4
SS:h450
allow direct media- Uncheck

Short code
4xxx
dial
.
LG-4

 

[IamaSherpa]

If it's a Server Edition system it automatically builds a SCN network, it means you don't need to create any short code for calls between extensions, remove it and try it again

 

[VinylDave]

Try using Monitor - Status - (S)RTP Sessions. This will show the RTP streams, RX and TX so at least you will know if it's a routing issue and where.

 

[phoneproblems]

Vinyldave- first time using this feature- pretty cool , please help explain the below./ or anyone
the trace results i received

Dst Ipaddress Site A,
(DST Port-40872) (Rx Octets 0)(Tx Octets 243040)(Pkts Sz 160)(Codec G711ulaw)(Bi-DIr 1)
( Rx Octets slow 0) (TX Octets slow 243040) (age 34598) (Delay 0) (Jitter 0) (Loss 0)
(remote jitter 0) (Remote loss 0)

Dst Ip address SIte B
(DST Port-40750) (Rx Octets 243520) (Tx Octets 5280) (Pkts Sz 160) (Codec G711ulaw) (Bi-DIr 1)
(Rx Octets slow 243520) (TX Octets slow 5280) (age 34850) (Delay 656) (Jitter 0) (Loss 0)
(remote jitter 0) (Remote loss 0)

 

[VinylDave]

Dst Ipaddress Site A,
(DST Port-40872) (Rx Octets 0)(Tx Octets 243040)
Site A is not receiving any voice packets but is sending ok.

Dst Ip address SIte B
(DST Port-40750) (Rx Octets 243520) (Tx Octets 5280)
Site B is both sending and receiving.

So you only have an issue on Site A receiving voice.
Assuming your routing is correct what's in between site a and b, firewall?

 

[phoneproblems]

i dont hear audio on either side -

yes sonic wall FW

i had him uncheck h323,sip trans etc. open the necesary ports, approve site A traffic, no luck.

 

[VinylDave]

is there any packet inspection, SPI, going on? You need to remove RTP from these zones if configured. Other than that, it all points to the firewall in my opinion. What ports are open and are they open for both source and destination? You would expect with a vpn that all traffic was allowed, hence my point 1.

 

[IPOTS]

Have you tried to enable "Allow Direct Media Path"?
I see you have them unchecked. What happens if you enable them on both sides?
Also check your RTP ports are 40750-50750 on both systems. Sometimes this get changed and can cause issues with one way speech.

On the firewall ALG fixups must be disabled.

https://ipoffice-techsolutions.com/

 

[Westi]

It sounds like you use the public IP with forwarded ports and not a VPN connection, could you confirm that?

Joe
FHandw, ACSS, ACIS

https://www.millsidetc.com/

 

[phoneproblems]

IPOTS
yes and same problme when checked and yes on both 4075-50750 ialso experienced that issue several month back- confirmed it this time , yes ALG disabled. i know sometimes a command on the FW needs to be disabled

 

[phoneproblems]

Westi
yes- FW is is whitelisted on the PUB from from site A & SITE B

the network diagram is as follow

SITE A - allow all traffic from SITE B PUB IP- pings bak forth no issues
SITE B is a Data center which routes traffic through a VPN tunnel from SITE B to SITE C and only the phones are connected to SITE C
( basically SITE B and C are VPN tunnel ) the IP Server abides in data center

Site A has its own site not connected via VPN tunnel by design for future separation within 2-3yrs.

i have allowed IP routes for all 0.0.0.0 and configured routes for direct subnet of xxx.xxx.xxx.0 point to lan 1 on both system at SITE A and B

 

[Westi]

That means that you most likely have a NAT'ing issue and your firewall settings (Server not firewall) are probably not working for you.
ICMP traffic is not a test for correct connectivity. Create a VPN and you are good to go rather than fiddling with the NAT issue

Joe
FHandw, ACSS, ACIS

https://www.millsidetc.com/

 

[VinylDave]

Yep, that's got NAT written all over it. Alternatively, set up a sip trunk between them if you can't vpn.