I have Asterisk 1.4.22.
I have to route call from Asterisk Server 1 to Asterisk Server 2.
I am getting this error.
When call come from Asterisk 1 to my Asterisk Box. it fails.
WARNING[4857]: chan_sip.c:8553 check_auth: username mismatch, have <8430735793>, digest has <73452591002>;tag=1957114284>
[Dec 22 05:36:49] NOTICE[4857]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user "73452591002"sip:73452591002;tag=1957114284
[Dec 22 05:36:59] WARNING[4857]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission SIPCALL-48871530-1933896248@XX.XXX.XX.XXX for seqno 2 (Critical Response) – See doc/sip-retransmit.txt.
for incoming calls context is defined so
SIP.CONF
[Incoming]
type=friend
host=XX.XX.XX.XX
context=from-incoming
disallow=all
allow=g729
allow=g723
allow=ulaw
allow=alaw
insecure=yes
nat=yes
EXNTENSIONS.CONF
[from-incoming]
exten => s,1,Wait,1
extem => s,2,Answer
exten => s,3,Wait,1
exten => _46685.,1,Dial(SIP/${EXTEN}@To-OutGoing)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => i,1,Playback(invalid)
Please Help