VoIP outgoing call problems

I am a new Asterisk and VoIP user and the pain is trying to determine who owns a problem. Is it me (Asterisk configuration), BroadVoice, or my DSL (2300 kbs down/700 kbs up) connection.

I’m seeing a couple intermittent problems…

(1) When a pbx sip phone is calling to an outside (TelCo) number, the pbx phone user continues to hear the ring tone even though the outside phone has been answered.

(2) When calls are connected (pbx to outside TelCo), inside the pbx hears excellent voice quality but the outside TelCo user says the voice quality is bad. Is this a function of the lower upload speed. I’ve checked when this occurs and the upload speed is around 700 kbs.

TIA for any help…

I experience a similar issue although in a different environment.

I have 160 voip based asterisk users (Koncept1020a and grandstream gpx2000) phones, a high speed lan (1gb) in well configured network and a fast linux box (asterisk uses less than 10% cpu on average) running fedora.
We use the TE110P pri interface cards on the telco interface and are configured as euro-isdn.

I use a slightly modified version of * 1.2 - no changes to any modules that handle the audio!
Changes include a voicemail type conf file for fax-to-email and modified app_rxfax to use it. A modified park call thread, to use retrydial instead of dial when returning to the original extension in park calls. Enhanced the manager channel status command to return a list of channels status by prefix for our own written soft operator switchboard.

Outbound audio sometimes deteriorates badly. Remote telco users can be heard clearly with no audio break-up by the asterisk voip user, but at the same time the remote part hears audio that is very choppy or breaks up. This does not happen on every call!

I have tried various codecs from g711 to gsm and all exhibit the problem.

I have ruled out our network as we run 600 call centre agents through our own home-grown intel dialogic based voip call centre system with excellent quality. The call centre agents soft-phone uses g723 codec.
Also the fact that I can hear the other party clearly at all times rules out bandwidth problem.

I have followed wiki and other sites audio problem resolution steps, including adjusting the rx and tx gains, I also as a last resort tried the different echo cancellations provided in * but I have not been able to resolve.

I suspect some incompatibility in the rtp frame sizes between the digium card and phones but I dont know how to verify this.

Any pointers?