Constant Voice issues with Asterisk

[Prof.Merlin]

Hi All

We are having constant issues with our asterisk box. We either have dropped calls, or a lot of one way voice.
Specs on the machine is 2 x X3430 @ 2.40ghz with 12GB memory. We are running 2 other services on the same box, but will be moving off soon.

We have tried 3 different connectivity options, now running with neotel 15mbit fibre. Pings run between 3ms-6ms to Centracom(our sip provider).

We have run the dahdi_test and got the following results:

--- Results after 435 passes ---
Best: 100.000 -- Worst: 99.612 -- Average: 99.993044, Difference: 99.997798

We are running with a Sangoma A200 card.

In our trunks, under outgoing, we have:
allow=g729

and incoming we have:
disallow=all
allow=G729&alaw&ulaw&gsm
qualify=yes

We have a digium G729 codec.

I am not sure what other info I can give.

Our phones are all set to use different codecs, would changing them all to G729 make a difference?

Thank you

 

[gmza]

one-way voice is usually a firewalling or NAT related issue, particularly when the other side can hear you but you cannot hear the other side.

I wouldn't recommend enabling GSM; the audio quality isn't great. G.729 is preferrable, however, there really should be no issue running A-law (or u-law, the US equivalent) given how much bandwidth you have at your disposal.

Regarding dropped calls, have you tried to see if you experience the same conditions to another VoIP provider? If you don't want to go that far, then what I can suggest is running a "sip set debug peer centracom" (or whatever the name of the trunk is) and watching the SIP packets on the dropped calls. Try and see if the call is teared down by your PBX or by the other side and if Asterisk logs any errors prior the call tear-down (e.g. complaints about packet retransmissions, timeouts, etc.)

 

[InfinityMVS]

Hi All

We are having constant issues with our asterisk box. We either have dropped calls, or a lot of one way voice.
Specs on the machine is 2 x X3430 @ 2.40ghz with 12GB memory. We are running 2 other services on the same box, but will be moving off soon.

We have tried 3 different connectivity options, now running with neotel 15mbit fibre. Pings run between 3ms-6ms to Centracom(our sip provider).

We have run the dahdi_test and got the following results:

--- Results after 435 passes ---
Best: 100.000 -- Worst: 99.612 -- Average: 99.993044, Difference: 99.997798

We are running with a Sangoma A200 card.

In our trunks, under outgoing, we have:
allow=g729

and incoming we have:
disallow=all
allow=G729&alaw&ulaw&gsm
qualify=yes

We have a digium G729 codec.

I am not sure what other info I can give.

Our phones are all set to use different codecs, would changing them all to G729 make a difference?

Thank you

Hi Prof,

I suggest the following:

1.Leave the incoming/user details sections blank.

2.Put the following in your outgoing / peer details

username=[your user name]
type=peer
secret=[your secret]
qualify=yes
nat=yes
insecure=very
host=[your sip provider host ip supplied to you]
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=yes
authname=[yourusername]
allow=g729

3.Check with your SP how they want the registration string usually [your user name]:[your secret]@[sip host ip]/[your user name]

As far as the IP phones are concerned if you disable all codecs except G729 it might cause other issues on your PSTN/Voicemal/inter-extension calls, rather leave it as is and allow the system to transcode from extension codec to G729.

If the dropped calls / one way voice is on a remote extension registering on the asterisk box via SIP, check the RTP settings on the remote phone and make them the same as what asterisk is expecting (/etc/asterisk/rtp.conf)

If you are still experiencing dropped calls it's time to look at your SP, ask Centracom exactly who they are passing the traffic to and if they have capacity as your description suggest that you have connectivity running to them, and from there on it breaks into the relevant SP's.

The sangoma card you mentioned is a analogue card, in most VoIP only cases this only provides timing to asterisk, and analogue PSTN connections to the Asterisk box

If you are still experiencing problems I can gladly assist you with a SiP only account to use as a test and see if you are experiencing the same issues.

Regards,


Eugene Meyer
021 200 1780

 

[Prof.Merlin]

Thanks for the replies. If i remove everything from incoming, would that not mean that i wont get any incoming calls?

 

[InfinityMVS]

if the sip trunk registers with the setting as above you will be able to make and receive calls.

 

[werny]

I HATE setting up SIP,IAX2 or ZAP Trunks!!! It's always a struggle to set it up correctly.

The codecs wouldn't matter, I know of a company that uses alot of different codecs and it still works great, they also bought they're own codecs.

 

[Prof.Merlin]

Think i figured out the issue. Added the nat and externip config and forwarded the rtp ports and it seemed ro have sorted the problem out.

Thanks for all your help.