Intermittent Problem with No/bad audio on incoming calls

[elkonrad]

Hi,

I am out of ideas. I have an Intermittent problem with some incoming calls. What happens is:

When person calls asterisk box via sip trunk directly to queue/ring group with fixed agents there is no audio or caller can hear some kind of breaks but cannot hear called person. When you hang up and call again, Call is perfectly fine. It think it happens after period of inactivity on the system ( I don't how long is inactivity). There is no problem with outgoing calls.

It happens on PBXIAF with asterisk 1.6.2.18 and asterisk 1.4.42
All boxes use ulaw/alaw. There is also cisco 887 router with ports 5060/5061 forwarded to asterisk. RTP is to be handled by ipinspect.

All help very welcome.

Thanks,
K.

 

[jmullinix]

What is ipinspect? You should have UDP ports 10000-20000 forwarded to your phone system also.

 

[elkonrad]

What is ipinspect? You should have UDP ports 10000-20000 forwarded to your phone system also.

Hi,
This is also happening on the box with all ports forwarded to the pbx. I have ASA5505 in that case, nat config below.


static (HostedPABX,outside) XX.XX.XX.XX 192.168.176.2 netmask 255.255.255.255

All these ports permitted:

access-list outside_access_in extended permit udp any host XX.XX.XX.XX eq 4569
access-list outside_access_in extended permit udp any host XX.XX.XX.XX eq sip
access-list outside_access_in extended permit tcp any host XX.XX.XX.XX eq sip
access-list outside_access_in extended permit tcp any host XX.XX.XX.XX eq 5061
access-list outside_access_in extended permit udp any host XX.XX.XX.XX object-group PBX_RTP
object-group service PBX_RTP udp
port-object range 10000 20000

 

[atsak]

Do you have SIP fixup (ie ALG) enabled?

 

[elkonrad]

Do you have SIP fixup (ie ALG) enabled?

Hi,

It was enabled but I disabled that and did not make any difference. Bad call still happens. When you hangup and call again it is perfect.

K.