<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><HTML DIR=ltr><HEAD><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1"></HEAD><BODY><DIV><FONT face='Arial' color=#000000 size=2>Hi All</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I seem to have a small problem with the music on
hold button on SJPhone.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have 2 asterisk installations one from the Rapid
distribution and one from the latest CVS.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>On the rapid dist when I press the music on hold
button on my SJPhone I get music on hold.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>When I do the same I get no music on hold just
silence.</FONT></DIV>
<DIV><FONT face=Arial size=2>I create extension like this exten =>
1111,1,MusicOnHold(Default), and when I dial it then I hear music, so music on
hold works but the hold button do not.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Can anyone help with this?</FONT></DIV>
<DIV><FONT face=Arial size=2> is this a bug in CVS?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>here are debugs from both installs (1 working and 1
not working):</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>********************** WORKING
************************</FONT></DIV>
<DIV><FONT face=Arial size=2>Sip read:<BR>INVITE sip:asterisk@xxx.xxx.xxx.xxx
SIP/2.0<BR>l: 214<BR>m: <sip:4802@192.168.1.111:5060><BR>i: <A
href="mailto:08c50c2469d676562285f02f72e5f6be@xxx.xxx.xxx.xxx">08c50c2469d676562285f02f72e5f6be@xxx.xxx.xxx.xxx</A><BR>c:
application/sdp<BR>Max-Forwards: 70<BR>CSeq: 13 INVITE<BR>f:
<sip:4802@xxx.xxx.xxx.xxx:2841>;tag=41280171719448<BR>t:
<sip:asterisk@xxx.xxx.xxx.xxx>;tag=as7cf27066<BR>User-Agent:
SJLabs-SJphone/1.30.252<BR>v: SIP/2.0/UDP
192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000078</FONT></DIV>
<P><FONT face=Arial size=2>v=0<BR>o=- 3318544820 3318544833 IN IP4
192.168.1.111<BR>s=SJphone<BR>c=IN IP4 0.0.0.0<BR>t=0
0<BR>a=direction:active<BR>m=audio 16394 RTP/AVP 3 101<BR>a=rtpmap:3
GSM/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-11,16</FONT></P>
<P><FONT face=Arial size=2>11 headers, 10 lines<BR>Using latest request as basis
request<BR>Sending to 192.168.1.111 : 5060 (NAT)<BR>Found audio format
UNKN<BR>Found audio format UNKN<BR>Found description format GSM<BR>Found
description format telephone-event<BR>Capabilities: us - 6, them - 2/0, combined
- 2<BR>Non-codec capabilities: us - 1, them - 1, combined - 1<BR>We're
at xxx.xxx.xxx.xxx port 14276<BR>Answering with preferred capability
2<BR>Answering with non-codec capability 1<BR>Reliably Transmitting
(NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000078;received=xxx.xxx.xxx.xxx<BR>From:
<sip:4802@xxx.xxx.xxx.xxx:2841>;tag=41280171719448<BR>To:
<sip:asterisk@xxx.xxx.xxx.xxx>;tag=as7cf27066<BR>Call-ID: <A
href="mailto:08c50c2469d676562285f02f72e5f6be@xxx.xxx.xxx.xxx">08c50c2469d676562285f02f72e5f6be@xxx.xxx.xxx.xxx</A><BR>CSeq:
13 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:asterisk@xxx.xxx.xxx.xxx><BR>Content-Type:
application/sdp<BR>Content-Length: 219</FONT></P>
<P><FONT face=Arial size=2>v=0<BR>o=root 17002 17015 IN IP4
xxx.xxx.xxx.xxx<BR>s=session<BR>c=IN IP4 195.216.65.216<BR>t=0 0<BR>m=audio
14276 RTP/AVP 3 101<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - - -</FONT></P>
<P><FONT face=Arial size=2> to xxx.xxx.xxx.xxx:2841<BR></FONT></P>
<P><FONT face=Arial size=2></FONT> </P>
<P><FONT face=Arial size=2>*********************** NOT WORKING
********************************</FONT></P>
<P><FONT face=Arial size=2>Sip read:<BR>INVITE sip:4803@192.168.1.20
SIP/2.0<BR>l: 214<BR>m: <sip:4803@192.168.1.111:5060><BR>i: <A
href="mailto:0b36753944573aa4681709356c705397@192.168.1.20">0b36753944573aa4681709356c705397@192.168.1.20</A><BR>c:
application/sdp<BR>Max-Forwards: 70<BR>CSeq: 1 INVITE<BR>f:
<sip:4803@192.168.1.111:5060>;tag=41308811925234<BR>t:
<sip:4803@192.168.1.20>;tag=as463b04a6<BR>User-Agent:
SJLabs-SJphone/1.30.252<BR>v: SIP/2.0/UDP
192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000000c8</FONT></P>
<P><FONT face=Arial size=2>v=0<BR>o=- 3318545106 3318545107 IN IP4
192.168.1.111<BR>s=SJphone<BR>c=IN IP4 0.0.0.0<BR>t=0
0<BR>a=direction:active<BR>m=audio 16400 RTP/AVP 3 101<BR>a=rtpmap:3
GSM/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-11,16</FONT></P>
<P><FONT face=Arial size=2>11 headers, 10 lines<BR>Using latest request as basis
request<BR>Sending to 192.168.1.111 : 5060 (NAT)<BR>We're at 192.168.1.20 port
18336<BR><FONT color=#ff0000>Answering/Requesting with root capability 0x4
(ulaw)<BR>Answering with preferred capability 0x2 (gsm)<BR>Answering with
non-codec capability 0x1 (telephone-event)</FONT><BR>Reliably Transmitting
(NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
192.168.1.111;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000000c8;received=192.168.1.111;rport=5060<BR>From:
<sip:4803@192.168.1.111:5060>;tag=41308811925234<BR>To:
<sip:4803@192.168.1.20>;tag=as463b04a6<BR>Call-ID: <A
href="mailto:0b36753944573aa4681709356c705397@192.168.1.20">0b36753944573aa4681709356c705397@192.168.1.20</A><BR>CSeq:
1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:4803@192.168.1.20><BR>Content-Type:
application/sdp<BR>Content-Length: 241</FONT></P>
<P><FONT face=Arial size=2>v=0<BR>o=root 12791 12793 IN IP4
192.168.1.111<BR>s=session<BR>c=IN IP4 192.168.1.111<BR>t=0 0<BR>m=audio 16398
RTP/AVP 0 3 101<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - - -</FONT></P>
<P><FONT face=Arial size=2> to 192.168.1.111:5060<BR></FONT></P>
<P>Thanks</P>
<P>KF</P></BODY></HTML>