Call is terminating after few seconds and one way audio

Asterisk Asterisk Support
1

Hi,

I am using Asterisk 11.7 and the OS is Ubuntu 12.04.

My Server IP address: 5.5.5.9 and all phones are connected to this address with in the LAN only and able to make/receive calls successfully.

Below is my network architecture:

ISP Cable -> Normal Router (Global IP Address) -> LAN Switch -> Various Phones

Now, I have done the port forwarding (5060 to 5.5.5.9) in my router and able to register with the Asterisk server from outside. When I try to make calls with in the extensions and as well as outside PSTN numbers also, call is connecting and disconnecting automatically after few seconds. Also, getting one way audio only.

Please find the below configuration of sip.conf:

[general]
bindaddr=0.0.0.0
bindport=5060
disallow=all
allow=all
rtcachefriends=yes
qualify=5000
rtsavesysname=yes
rtupdate=yes
ignoreregexpire=yes

CLI Messages:

-- DAHDI/i1/9123456789-1 is ringing
-- DAHDI/i1/9123456789-1 answered SIP/1001-00000000

[Apr 4 16:32:10] WARNING[27404]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 6399ms with no response
[Apr 4 16:32:10] WARNING[27404]: chan_sip.c:4204 retrans_pkt: Hanging up call wiki.asterisk.org/wiki/display/ … nsmissions).
– Hungup ‘DAHDI/i1/9123456789-1’

I hope this information helps to investigate the issue. Please do needful.

Thanks in advance.

Regards,
Mouli

2

You haven’t told asterisk what the public address is, or how to discover it.

Also, don’t forget to port forward the RTP ports, and you probably want to set allowguest to no.

3

This options could help you, change the values for your config

allowguest=no
externip=186.150.200.57
media_address=186.150.200.57
localnet=192.168.1.0/255.255.255.0

4

Hello,

As you suggested, I have added allowguest to no and global ip address with local net details. Now, it is working fine.

Thank you.

Regards,
Mouli