[Asterisk-Users] Help Me - SIP Phones ( No Voice) !!!!

[Jefferson Carvalho]

Hello list,

I've posted my problem on BSD list and i still have the
problem.
The remote side receives the call , but there's no voice
on the call.
I tried everything about possible NAT problems ..
but ther're on same net.

My platform:

FreeBSD 5.2.1-Release
Asterisk 1.0-RC2
soft phones : X-Lite

 >>>>
-- Executing Dial("SIP/1260-a7ae", "SIP/1262|20") in new stack
-- Called 1262
-- SIP/1262-c597 is ringing
-- SIP/1262-c597 answered SIP/1260-a7ae
-- Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
- Attempting native bridge of SIP/1260-a7ae and SIP/1262-c597
Sep  1 14:53:17 WARNING[135442432]: chan_sip.c:673 retrans_pkt: Maximum 
retries exceeded on call 
DB93109A-FC24-11D8-B3A5-005004803F0B at 172.20.1.133 for seqno 11288 
(Non-critical Response)
*
 >>>>> My sip.conf

*[1260]
type=friend
username=1260
secret=jeff
context=sip
qualify=300
mailbox=1260
callerid="Jefferson Carvalho" <1260>
host=dynamic
nat=no
canreinvite=no
allow=gsm
;
[1262]
type=friend
context=sip
username=1262
secret=1262
qualify=300
callerid="Ialle" <1262>
host=dynamic
nat=no
canreinvite=no
allow=gsm
;

*>> My extensions.conf
*
[general]

static=yes
writeprotect=no

[globals]

CONSOLE => Console/dsp
IAXINFO => guest
TRUNK => Zap/g2
TRUNKMSD => 1

[sip]

exten => 1260,1,Dial(SIP/1260,20)
exten => 1261,1,Dial(SIP/1261,20)
exten => 1262,1,Dial(SIP/1262,20)

Best Regards,

-Jefferson Carvalho
 IT Analist
 Credishop S/A
 Teresina-PI-Brasil
 5586-94321901