<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2900.2722" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>Dear all, </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I am getting the below errors when using asterisk.
I am using sjphone for testing purpose.</FONT></DIV>
<DIV><FONT face=Arial size=2>Below are the setting for sip.conf and
extension.conf</FONT></DIV>
<DIV><FONT face=Arial size=2>When i dial the number it rings on the remote
telephone. but after ringing 1 time it will disconnect.</FONT></DIV>
<DIV><FONT face=Arial size=2>Can anybody tell me what does this error means and
the how to solve this issue.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanking You,</FONT></DIV>
<DIV><FONT face=Arial size=2>Joel</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>sip.conf</FONT></DIV>
<DIV><FONT face=Arial size=2>[general]</FONT></DIV>
<DIV><FONT face=Arial size=2>context=default</FONT></DIV>
<DIV><FONT face=Arial size=2>port=5060</FONT></DIV>
<DIV><FONT face=Arial size=2>binaddr=0.0.0.0</FONT></DIV>
<DIV><FONT face=Arial size=2>srvlookup=yes</FONT></DIV>
<DIV><FONT face=Arial size=2>disallow=all</FONT></DIV>
<DIV><FONT face=Arial size=2>allow=g729</FONT></DIV>
<DIV><FONT face=Arial size=2>allow=g723</FONT></DIV>
<DIV><FONT face=Arial size=2>allow=ulaw</FONT></DIV>
<DIV><FONT face=Arial size=2>allow=ilbc</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[voip]</FONT></DIV>
<DIV><FONT face=Arial size=2>type=peer</FONT></DIV>
<DIV><FONT face=Arial size=2>host=202.202.202.202</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>and here is the extension.conf. I have placed in
the middle of extension.conf</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>exten => _X.,1,Dial(<A
href="mailto:SIP/${EXTEN}@voip">SIP/${EXTEN}@voip</A>)<BR>exten =>
_X.,2,Hangup<BR></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>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)<BR>Aug 11 10:15:02 NOTICE[11260]: channel.c:1736
ast_set_read_format: Unable to find a path from g723 to g729<BR>Aug 11 10:15:02
NOTICE[11260]: channel.c:1703 ast_set_write_format: Unable to find a path from
g729 to g723<BR> -- SIP/isphone-8213 is making progress
passing it to SIP/200-1264<BR>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)<BR>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)<BR>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)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT><FONT face=Arial size=2>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)<BR> -- SIP/isphone-8213
answered SIP/200-1264<BR>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)<BR>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<BR> == Spawn extension (default, 14025695651, 1) exited
non-zero on 'SIP/200-1264'<BR>ast*CLI> <BR>ast*CLI> <BR></DIV>
<DIV><BR></DIV></FONT></BODY></HTML>