SIP problem:No comtatible codecs

Asterisk Asterisk Support
1

hello, asterisk guys:
I use druid to make a call between two xlite phones. the error is as follow:
Process_sdp:No comtatible codecs
anyone knows that problem?
thanks
zhu8080

2

what codecs are you setting for each peer/user, and what have you set in sip.conf ? a log file fragment for a failed call would be useful too.

3

#include druid-trunk-sip.conf
[general]
context=druid-trunks-inbound
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[authentication]

[900]
type=friend
allow=ulaw
allow=alaw
allow=gsm
allow=g729
username=900
secret=123456
host=dynamic
context=druid-users
callerid=“900” <900>
amaflags=default
dtmfmode=rfc2833
video=no
restrictcid=no
insecure=very
mailbox=900
canreinvite=yes
nat=yes
qualify=yes
defaultip=192.168.2.159
disallow=all
fromuser=900
fromdomain=192.168.2.159

[800]
type=friend
allow=gsm
allow=ulaw
allow=alaw
allow=g729
username=800
secret=123456
host=dynamic
context=druid-users
callerid=“800” <800>
amaflags=default
video=no
restrictcid=no
insecure=very
canreinvite=yes
nat=no
qualify=yes
defaultip=192.168.2.155
disallow=all
fromuser=800
fromdomain=192.168.2.155
dtmfmode=rfc2833
the log is here:
Using latest REGISTER request as basis request
Sending to 192.168.2.155 : 21128 (NAT)
Transmitting (no NAT) to 192.168.2.155:21128:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.155:21128;branch=z9hG4bK-d87543-ce5e420df462a158-1–d87543-;received=192.168.2.155;rport=21128
From: "800"sip:800@192.168.2.138;tag=e97daa61
To: "800"sip:800@192.168.2.138
Call-ID: Y2FjZGQxMWNhZThkMjk5NTMzMzA1NDBkNWUzMDEzNGY.
CSeq: 560 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:800@192.168.2.138
Content-Length: 0

Transmitting (no NAT) to 192.168.2.155:21128:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.155:21128;branch=z9hG4bK-d87543-ce5e420df462a158-1–d87543-;received=192.168.2.155;rport=21128
From: "800"sip:800@192.168.2.138;tag=e97daa61
To: "800"sip:800@192.168.2.138;tag=as454f747a
Call-ID: Y2FjZGQxMWNhZThkMjk5NTMzMzA1NDBkNWUzMDEzNGY.
CSeq: 560 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 20
Contact: sip:800@192.168.2.155:21128;rinstance=7e40563c9c4403bc;expires=20
Date: Wed, 14 Mar 2007 13:11:40 GMT
Content-Length: 0
thanks

4

that’s not the log is it ? turn on debug logging in logger.conf, place the call and post the output to the log file here. there’s probably no need to have “sip debug” on at this point,

5

Hi

It looks as is you have disallowed all codecs.

The disallow must be first not last.

[900]
type=friend
disallow=all ; ADD THIS HERE
allow=ulaw
allow=alaw
allow=gsm
allow=g729
username=900
secret=123456
host=dynamic
context=druid-users
callerid=“900” <900>
amaflags=default
dtmfmode=rfc2833
video=no
restrictcid=no
insecure=very
mailbox=900
canreinvite=yes
nat=yes
qualify=yes
defaultip=192.168.2.159
; disallow=all REMOVE THIS
fromuser=900
fromdomain=192.168.2.159

6

hello, asteriskers:
i changed the disallow to top. it works. the two xlite can call each other. i will do more testing.
thanks!
zhu8080