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<DIV><FONT face=Arial size=2>more about the same problem ...</FONT></DIV>
<DIV><FONT face=Arial size=2>i've been playing around and got to this error
message which seems relevant ..</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>*CLI> dial 1303<BR> --
Executing Dial("OSS/dsp", "<A
href="mailto:SIP/723@216.52.153.207">SIP/723@216.52.153.207</A>") in new
stack<BR> -- Called <A
href="mailto:723@216.52.153.207">723@216.52.153.207</A><BR> --
SIP/216.52.153.207-1fb9 answered OSS/dsp<BR> << Console call has been
answered >><BR>NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read):
Unknown RTP codec 19 received<BR>NOTICE[1232188736]: File rtp.c, Line 326
(ast_rtp_read): Unknown RTP codec 19 received<BR>NOTICE[1232188736]: File rtp.c,
Line 326 (ast_rtp_read): Unknown RTP codec 19 received<BR>NOTICE[1232188736]:
File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19
received<BR>Killed<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>am I right in thinking i need a different codec to
connect to the sip host I want to</FONT></DIV>
<DIV><FONT face=Arial size=2>connect to? where do codecs come from?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>many cheers</FONT></DIV>
<DIV><FONT face=Arial size=2>Dave</FONT></DIV>
<DIV><FONT face=Arial size=2> </DIV></FONT>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=dtoma@fx.ro href="mailto:dtoma@fx.ro">Dan</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, May 30, 2003 7:50 PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] a
beginner's SIP question ..</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>Hi Dave,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>If you have registered the SIP phone with
Asterisk, then you must have a line like:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>exten => 555,1,dial(<A
href="mailto:SIP/723@216,52,153.207">SIP/723@216,52,153.207</A>)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>in extensions.conf file</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Then call 555 from the SIP phone to access the
destination.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>BR,</FONT></DIV>
<DIV><FONT face=Arial size=2>Dan</FONT></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=david@melita.net href="mailto:david@melita.net">Dave Alan
Caruana</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, May 30, 2003 6:21
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] a
beginner's SIP question ..</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>I have included a dump of the debug info
...</FONT></DIV>
<DIV><FONT face=Arial size=2>what I am trying to do is route a call from
sipphone 217.168.168.49</FONT></DIV>
<DIV><FONT face=Arial size=2>through asterisk 217.168.168.51 onto a gateway
<A href="mailto:723@216.52.153.207">723@216.52.153.207</A></FONT></DIV>
<DIV><FONT face=Arial size=2>If i dial direct from the sip phone to the
gateway it works fine .. so </FONT></DIV>
<DIV><FONT face=Arial size=2>I do not think there is any incompatibility
there.</FONT></DIV>
<DIV><FONT face=Arial size=2>Calls don't go through though ...</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>please help!!!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>cheers</FONT></DIV>
<DIV><FONT face=Arial size=2>Dave</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>*CLI> -- Executing
Dial("SIP/217.168.168.49:5060", "<A
href="mailto:SIP/723@216.52.153.207">SIP/723@216.52.153.207</A>") in new
stack<BR> -- Called <A
href="mailto:723@216.52.153.207">723@216.52.153.207</A><BR>
-- SIP/216.52.153.207-eca2 answered
SIP/217.168.168.49:5060<BR> -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-eca2<BR>WARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call <A
href="mailto:call-1054307890-9@217.168.168.49">call-1054307890-9@217.168.168.49</A>
for seqno 1 (Response)<BR> == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'<BR>WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call <A
href="mailto:call-1054307890-9@217.168.168.49">call-1054307890-9@217.168.168.49</A>
for seqno 1 (Response)<BR> -- Executing
Dial("SIP/217.168.168.49:5060", "<A
href="mailto:SIP/723@216.52.153.207">SIP/723@216.52.153.207</A>") in new
stack<BR> -- Called <A
href="mailto:723@216.52.153.207">723@216.52.153.207</A><BR>
-- SIP/216.52.153.207-1418 answered
SIP/217.168.168.49:5060<BR> -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-1418<BR>WARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call <A
href="mailto:call-1054307890-9@217.168.168.49">call-1054307890-9@217.168.168.49</A>
for seqno 1 (Response)<BR> == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'<BR>WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call <A
href="mailto:call-1054307890-9@217.168.168.49">call-1054307890-9@217.168.168.49</A>
for seqno 102 (Request)<BR> -- Executing
Dial("SIP/217.168.168.49:5060", "<A
href="mailto:SIP/723@216.52.153.207">SIP/723@216.52.153.207</A>") in new
stack<BR> -- Called <A
href="mailto:723@216.52.153.207">723@216.52.153.207</A><BR>
-- SIP/216.52.153.207-11ed answered
SIP/217.168.168.49:5060<BR> -- Attempting native bridge of
SIP/217.168.168.49:5060 and SIP/216.52.153.207-11ed<BR>WARNING[1125329600]:
File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call <A
href="mailto:call-1054307890-9@217.168.168.49">call-1054307890-9@217.168.168.49</A>
for seqno 1 (Response)<BR> == Spawn extension (default, 1303, 1)
exited non-zero on 'SIP/217.168.168.49:5060'<BR>WARNING[1125329600]: File
chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call <A
href="mailto:call-1054307890-9@217.168.168.49">call-1054307890-9@217.168.168.49</A>
for seqno 102 (Request)<BR></DIV></FONT>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=dtoma@fx.ro href="mailto:dtoma@fx.ro">Dan</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, May 29, 2003 8:15
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] a
beginner's SIP question ..</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV>
<DIV><FONT face=Arial size=2>Check to have a common set of
codecs.</FONT></DIV>
<DIV><FONT face=Arial size=2>If X-Lite is used and at the other end is a
phone without GSM support, then it doesn't work.</FONT></DIV>
<DIV><FONT face=Arial size=2>Try to disable GSM on the soft phone (if
X-Lite).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>BR,</FONT></DIV>
<DIV><FONT face=Arial size=2>Dan</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=david@melita.net href="mailto:david@melita.net">Dave Alan
Caruana</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, May 29, 2003 9:01
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] a
beginner's SIP question ..</DIV>
<DIV><FONT face=Arial size=2></FONT><FONT face=Arial
size=2></FONT><BR></DIV>
<DIV><FONT face=Arial size=2>I am trying to get asterisk to dial this
address :</FONT></DIV>
<DIV><FONT face=Arial size=2>sip:723@216.52.153.207</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Using a softphone on my PC
(217.168.168.49)</FONT></DIV>
<DIV><FONT face=Arial size=2>it dials immediately and I get a voice
prompt ..</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have configured an extension, 1303 on
asterisk,</FONT></DIV>
<DIV><FONT face=Arial size=2>modifying the demo configuration
:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>exten => 1303,1,Dial(<A
href="mailto:SIP/723@216.52.153.207">SIP/723@216.52.153.207</A>)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>When from my softphone I dial</FONT></DIV>
<DIV><FONT face=Arial size=2>sip:1303@217.168.168.51</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>on the console I get :</FONT></DIV>
<DIV><FONT face=Arial size=2> -- Executing
Dial("SIP/sipphone-97b6", "<A
href="mailto:SIP/723@216.52.153.207">SIP/723@216.52.153.207</A>") in new
stack<BR> -- Called <A
href="mailto:723@216.52.153.207">723@216.52.153.207</A><BR>
-- SIP/216.52.153.207-7c3b answered
SIP/sipphone-97b6<BR> -- Attempting native bridge of
SIP/sipphone-97b6 and SIP/216.52.153.207-7c3b</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>but on my headset all I get is silence ..
the call doesn't drop though.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>What am I doing wrong ?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>many thanks,</FONT></DIV>
<DIV><FONT face=Arial size=2>Dave</FONT></DIV>
<DIV><FONT face=Arial
size=2></FONT> </DIV></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>