Hi,
I’m having consistent problems making a conference call with my Grandstream GXP2000 phone and Asterisk 1.4.4.
Use Case:
- On the phone, I dial one number, put it on hold
- Dial the second number
- Push the phone “conference button”
- The call looks to be conferenced. However, I can only hear sound from one number and not the other.
(basic config details: sip, provider is Vitelity, g729 codec required for provder connection).
Apparently, this is due to problems the use of the g729 codec. When I use “sip show channels”, I see 5 lines
local extension g729
remote number g729
local extension g729
other remote number g729
local extension ulaw
Before I hit the “conference” button, I just see the first 4 lines, all g729. When i conference, I see the ulaw line. My guess is that there’s no conversion of sound between g729 and ulaw.
When I check “show g729” it tells me I have 4 licensed channels.
As an experiment, I reconfigured the Grandstream phone to only allow g729. (right now it allows g729a, PCMU, PCMA, GSM). Now, when I click the conference button, it hangs up one of the lines, and I see an error in Asterisk
chan_sip.c: No compatible codecs, not accepting this offer!
Any suggestions? Excerpts from sip.conf below.
[general]
...
; codecs
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
[90]
mailbox=90
username=xx
secret=xx
type=friend
host=dynamic
context=internal-forio
dtmfmode=rfc2833
canreinvite=no
[vitel-outbound]
type=friend
host=outbound1.vitelity.net
context=outbound-to-vitelity
username=xx
fromuser=xx
trustrpid=yes
sendrpid=yes
secret=xx
disallow=all
allow=g729