[Asterisk-Users] Sipura line 1 outgoing voice problem?

Senad Jordanovic senad at boltblue.com
Wed Mar 17 03:58:59 MST 2004

Chris Higgins wrote:
> Back in January I started having a problem with my Sipura (and there
> was 
> at least one other on the list with the same problem) that if I answer
> an incoming call (via X100P) on line 1 of my Sipura, the caller cannot
> hear any voice from the internal extension.  If the internal user puts
> the external user on hold (via flash hook) and returns, both
> directions 
> of audio are fine.

I have not had this problem... And I use X100P as well in same setup.
BUT... There are other problems I have or had.

> [cordless1]
> type=friend
> username=cordless1
> secret=xxx
> host=dynamic
> context=cordless1
> dtmfmode=info
> mailbox=1234
> canreinvite=no
> disallow=all
> allow=alaw

If you are using your SPA 2000 directly with * maybe it is better to
have "canreinvite" set to yes. ??? 

Also.. I think that auto default dtmfmode for SPA is AVT (which is
RFC2833)... So check that.!!!

More information about the asterisk-users mailing list