[Asterisk-Users] Video phone failed on Asterisk-1.2.4

蘇玉華 yhs at cht.com.tw
Wed Mar 22 22:54:27 MST 2006


Dear sir,

I got trouble on InnoMedia video phone with Asterisk-1.2.4.
If InnoMedia video phone as a caller, then the call will be a success, no any problem.

The problem happens:
If InnoMedia video phone to be a callee, the call can not make successfully.
For instance, 
Caller 23267668 dialed to callee 23267663, callee 23267663 was ringing.
If I picked up the callee's phone, then abnormal situation happened 
and caller 23267668 will receive a '603 Declined' response finally.
( caller can be any kinds of sip phone )
 
My sip.conf: 
[23267663]                  ;InnoMedia video phone
type=friend          
username=23267663           
host=dynamic                ;10.144.169.133
port=5060    
context=default
;canreinvite=no
canreinvite=yes
language=cn 
disallow=all
allow=ulaw
allow=h263
allow=h263p
 
[23267668]               ;IA sip phone
type=friend
username=23267668
host=dynamic             ;10.144.169.138
port=5060
callgroup=3
context=default
pickupgroup=3
;canreinvite=no
canreinvite=yes
disallow=all
allow=ulaw

My extension.conf:
exten => 23267663,1,Dial(SIP/23267663,20,Ttr)
;exten => 23267663,1,Dial(SIP/23267663,20)
exten => 23267663,2,Hangup

*CLI>
Executing Dial("SIP/23267668-4459", "SIP/23267663|20|Ttr") in new stack
    -- Called 23267663
    -- Nobody picked up in 20000 ms
    -- Executing Hangup("SIP/23267668-4459", "") in new stack
  == Spawn extension (default, 23267663, 2) exited non-zero on 'SIP/23267668-445
9'
    -- Executing Hangup("SIP/23267668-4459", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/23267668-4459'
Mar 23 12:48:22 ERROR[26808]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql: cannot
 connect to database server localhost.
Mar 23 12:48:42 WARNING[23712]: chan_sip.c:1210 retrans_pkt: Maximum retries exc
eeded on transmission 221e15add42e8e26707990325527b6e5 at 10.144.169.138 for seqno
1145843625 (Critical Response)
*CLI>
     
I ever tried many combination of parameters in sip.conf, but did not help at all.
Can any one give me a help?   Thanks a lot.

By the way, it works fine using same configuration in Aterisk-1.0.9.


yuh-su
Chunghwa Telecom. Co., Ltd.
Telecommunication Laboratories
Taiwan, R.O.C 
email: yhs at cht.com.tw
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060322/5730834f/attachment.htm


More information about the asterisk-users mailing list