Dear all,
 
I am getting the below errors when using asterisk. I am using sjphone for testing purpose.
Below are the setting for sip.conf and extension.conf
When i dial the number it rings on the remote telephone. but after ringing 1 time it will disconnect.
Can anybody tell me what does this error means and the how to solve this issue.
 
Thanking You,
Joel
 
sip.conf
[general]
context=default
port=5060
binaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=g729
allow=g723
allow=ulaw
allow=ilbc
 
[voip]
type=peer
host=202.202.202.202
 
and here is the extension.conf. I have placed in the middle of extension.conf
 
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten => _X.,2,Hangup
 
Aug 11 10:15:01 WARNING[11260]: channel.c:2127 ast_channel_make_compatible: No path to translate from SIP/isphone-8213(256) to SIP/200-1264(4)
Aug 11 10:15:02 NOTICE[11260]: channel.c:1736 ast_set_read_format: Unable to find a path from g723 to g729
Aug 11 10:15:02 NOTICE[11260]: channel.c:1703 ast_set_write_format: Unable to find a path from g729 to g723
    -- SIP/isphone-8213 is making progress passing it to SIP/200-1264
Aug 11 10:15:02 WARNING[11260]: chan_sip.c:1836 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 256/256)
Aug 11 10:15:02 WARNING[11260]: chan_sip.c:1836 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 256/256)
Aug 11 10:15:02 WARNING[11260]: chan_sip.c:1836 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 256/256)
Aug 11 10:15:06 WARNING[11260]: chan_sip.c:1836 sip_write: Asked to transmit frame type 1, while native formats is 4 (read/write = 4/4)
    -- SIP/isphone-8213 answered SIP/200-1264
Aug 11 10:15:06 WARNING[11260]: channel.c:2127 ast_channel_make_compatible: No path to translate from SIP/200-1264(4) to SIP/isphone-8213(1)
Aug 11 10:15:06 WARNING[11260]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't make SIP/200-1264 compatible with SIP/isphone-8213
  == Spawn extension (default, 14025695651, 1) exited non-zero on 'SIP/200-1264'
ast*CLI>
ast*CLI>

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