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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
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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