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<DIV><FONT face=Arial size=2>I too am having the same problem with CVS from last
night. From my debugging, * never attempts to start MOH. Anyone else found
this?</FONT></DIV>
<BLOCKQUOTE
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=krystian.filiks@kfiliks.com
href="mailto:krystian.filiks@kfiliks.com">Krystian Filiks</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> Monday, February 28, 2005 1:46
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] music on hold
trouble</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2 ="olor=#000000">Hi All</FONT></DIV>=DIV><FONT
face=Arial size=2></FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2>I seem to have a small problem with
the =usic on =old button on SJPhone.</FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2></FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2>I have 2 asterisk installations one
=rom the Rapid =istribution and one from the latest CVS.</FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2></FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2>On the rapid dist when I press the
=usic on hold =utton on my SJPhone I get music on hold.</FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2></FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2>When I do the same I get no music
on =old just =ilence.</FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2>I create extension like this exten
=3D> =111,1,MusicOnHold(Default), and when I dial it then I hear music, so
=usic on =old works but the hold button do not.</FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2></FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2>Can anyone help with this?</FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2> is this a bug in CVS?</FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2></FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2></FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2>here are debugs from both installs
(1 =orking and 1 =ot working):</FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2></FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2>**********************
WORKING =***********************</FONT>
<DIV></DIV>=DIV><FONT face=Arial size=2>Sip read:<BR>INVITE
=ip:asterisk@xxx.xxx.xxx.xxx =IP/2.0<BR>l: 214<BR>m:
<sip:4802@192.168.1.111:5060><BR>i: <A
='ref="mailto:08c50c2469d676562285f02f72e5f6be@xxx.xxx.xxx.xxx"'>08c50c24=9d676562285f02f72e5f6be@xxx.xxx.xxx.xxx</A><BR>c:
=pplication/sdp<BR>Max-Forwards: 70<BR>CSeq: 13 INVITE<BR>f:
=lt;sip:4802@xxx.xxx.xxx.xxx:2841>;tag=41280171719448<BR>t:
=lt;sip:asterisk@xxx.xxx.xxx.xxx>;tag=as7cf27066<BR>User-Agent:
=JLabs-SJphone/1.30.252<BR>v: SIP/2.0/UDP
=92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000=78</FONT>
<DIV></DIV>=P><FONT face=Arial size=2>v=0<BR>o=- 3318544820 3318544833 IN
=P4 =92.168.1.111<BR>s=SJphone<BR>c=IN IP4 0.0.0.0<BR>t=0
=<BR>a=direction:active<BR>m=audio 16394 RTP/AVP 3 =01<BR>a=rtpmap:3
=SM/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 =-11,16</FONT>
<P></P>=P><FONT face=Arial size=2>11 headers, 10 lines<BR>Using latest
=equest as basis =equest<BR>Sending to 192.168.1.111 : 5060 (NAT)<BR>Found
audio format =NKN<BR>Found audio format UNKN<BR>Found description format
GSM<BR>Found =escription format telephone-event<BR>Capabilities: us - 6, them
- 2/0, =ombined = 2<BR>Non-codec capabilities: us - 1, them - 1, combined -
1<BR>We're =t xxx.xxx.xxx.xxx port 14276<BR>Answering with preferred
=apability =<BR>Answering with non-codec capability 1<BR>Reliably Transmitting
=NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
=92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000=78;received=xxx.xxx.xxx.xxx<BR>From:
=lt;sip:4802@xxx.xxx.xxx.xxx:2841>;tag=41280171719448<BR>To:
=lt;sip:asterisk@xxx.xxx.xxx.xxx>;tag=as7cf27066<BR>Call-ID: <A
='ref="mailto:08c50c2469d676562285f02f72e5f6be@xxx.xxx.xxx.xxx"'>08c50c24=9d676562285f02f72e5f6be@xxx.xxx.xxx.xxx</A><BR>CSeq:
=3 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, =PTIONS,
=YE, REFER<BR>Contact: =lt;sip:asterisk@xxx.xxx.xxx.xxx><BR>Content-Type:
=pplication/sdp<BR>Content-Length: 219</FONT>
<P></P>=P><FONT face=Arial size=2>v=0<BR>o=root 17002 17015 IN IP4
=xx.xxx.xxx.xxx<BR>s=session<BR>c=IN IP4 195.216.65.216<BR>t=0 =<BR>m=audio
=4276 RTP/AVP 3 101<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:101
=elephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - - =</FONT>
<P></P>=P><FONT face=Arial size=2> to =xx.xxx.xxx.xxx:2841<BR></FONT>
<P></P>=P><FONT face=Arial size=2></FONT>
<P></P>=P><FONT face=Arial size=2>*********************** NOT WORKING
=*******************************</FONT>
<P></P>=P><FONT face=Arial size=2>Sip read:<BR>INVITE sip:4803@192.168.1.20
=IP/2.0<BR>l: 214<BR>m: <sip:4803@192.168.1.111:5060><BR>i: <A
='ref="mailto:0b36753944573aa4681709356c705397@192.168.1.20"'>0b367539445=3aa4681709356c705397@192.168.1.20</A><BR>c:
=pplication/sdp<BR>Max-Forwards: 70<BR>CSeq: 1 INVITE<BR>f:
=lt;sip:4803@192.168.1.111:5060>;tag=41308811925234<BR>t:
=lt;sip:4803@192.168.1.20>;tag=as463b04a6<BR>User-Agent:
=JLabs-SJphone/1.30.252<BR>v: SIP/2.0/UDP
=92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227c5f0000648200000=c8</FONT>
<P></P>=P><FONT face=Arial size=2>v=0<BR>o=- 3318545106 3318545107 IN =P4
=92.168.1.111<BR>s=SJphone<BR>c=IN IP4 0.0.0.0<BR>t=0
=<BR>a=direction:active<BR>m=audio 16400 RTP/AVP 3 =01<BR>a=rtpmap:3
=SM/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 =-11,16</FONT>
<P></P>=P><FONT face=Arial size=2>11 headers, 10 lines<BR>Using latest
=equest as basis =equest<BR>Sending to 192.168.1.111 : 5060 (NAT)<BR>We're at
=92.168.1.20 port =8336<BR><FONT color=#ff0000>Answering/Requesting with root
capability =x4 =ulaw)<BR>Answering with preferred capability 0x2
(gsm)<BR>Answering =ith =on-codec capability 0x1
(telephone-event)</FONT><BR>Reliably =ransmitting =NAT):<BR>SIP/2.0 200
OK<BR>Via: SIP/2.0/UDP
=92.168.1.111;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000000c8;re=eived=192.168.1.111;rport=5060<BR>From:
=lt;sip:4803@192.168.1.111:5060>;tag=41308811925234<BR>To:
=lt;sip:4803@192.168.1.20>;tag=as463b04a6<BR>Call-ID: <A
='ref="mailto:0b36753944573aa4681709356c705397@192.168.1.20"'>0b367539445=3aa4681709356c705397@192.168.1.20</A><BR>CSeq:
= INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, =PTIONS,
=YE, REFER<BR>Contact: <sip:4803@192.168.1.20><BR>Content-Type:
=pplication/sdp<BR>Content-Length: 241</FONT>
<P></P>=P><FONT face=Arial size=2>v=0<BR>o=root 12791 12793 IN IP4
=92.168.1.111<BR>s=session<BR>c=IN IP4 192.168.1.111<BR>t=0 =<BR>m=audio 16398
=TP/AVP 0 3 101<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:3 =SM/8000<BR>a=rtpmap:101
=elephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - - =</FONT>
<P></P>=P><FONT face=Arial size=2> to 192.168.1.111:5060<BR></FONT>
<P></P>=P>Thanks
<P></P>=P>KF
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