[Asterisk-Users] music on hold trouble

Mon Feb 28 20:21:12 MST 2005

I too am having the same problem with CVS from last night. From my debugging, * never attempts to start MOH. Anyone else found this?
  ----- Original Message ----- 
  From: Krystian Filiks 
  To: asterisk-users at lists.digium.com 
  Sent: Monday, February 28, 2005 1:46 PM
  Subject: [Asterisk-Users] music on hold trouble

  Hi All
  =DIV> 
  =DIV>I seem to have a small problem with the =usic on =old button on SJPhone.
  =DIV> 
  =DIV>I have 2 asterisk installations one =rom the Rapid =istribution and one from the latest CVS.
  =DIV> 
  =DIV>On the rapid dist when I press the =usic on hold =utton on my SJPhone I get music on hold.
  =DIV> 
  =DIV>When I do the same I get no music on =old just =ilence.
  =DIV>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.
  =DIV> 
  =DIV>Can anyone help with this?
  =DIV> is this a bug in CVS?
  =DIV> 
  =DIV> 
  =DIV>here are debugs from both installs (1 =orking and 1 =ot working):
  =DIV> 
  =DIV>**********************  WORKING =***********************
  =DIV>Sip read:
  INVITE =ip:asterisk at xxx.xxx.xxx.xxx =IP/2.0
  l: 214
  m: <sip:4802 at 192.168.1.111:5060>
  i: 08c50c24=9d676562285f02f72e5f6be at xxx.xxx.xxx.xxx
  c: =pplication/sdp
  Max-Forwards: 70
  CSeq: 13 INVITE
  f: =lt;sip:4802 at xxx.xxx.xxx.xxx:2841>;tag=41280171719448
  t: =lt;sip:asterisk at xxx.xxx.xxx.xxx>;tag=as7cf27066
  User-Agent: =JLabs-SJphone/1.30.252
  v: SIP/2.0/UDP =92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000=78
  =P>v=0
  o=- 3318544820 3318544833 IN =P4 =92.168.1.111
  s=SJphone
  c=IN IP4 0.0.0.0
  t=0 =
  a=direction:active
  m=audio 16394 RTP/AVP 3 =01
  a=rtpmap:3 =SM/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 =-11,16

  =P>11 headers, 10 lines
  Using latest =equest as basis =equest
  Sending to 192.168.1.111 : 5060 (NAT)
  Found audio format =NKN
  Found audio format UNKN
  Found description format GSM
  Found =escription format telephone-event
  Capabilities: us - 6, them - 2/0, =ombined = 2
  Non-codec capabilities: us - 1, them - 1, combined - 1
  We're =t xxx.xxx.xxx.xxx port 14276
  Answering with preferred =apability =
  Answering with non-codec capability 1
  Reliably Transmitting =NAT):
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP =92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000=78;received=xxx.xxx.xxx.xxx
  From: =lt;sip:4802 at xxx.xxx.xxx.xxx:2841>;tag=41280171719448
  To: =lt;sip:asterisk at xxx.xxx.xxx.xxx>;tag=as7cf27066
  Call-ID: 08c50c24=9d676562285f02f72e5f6be at xxx.xxx.xxx.xxx
  CSeq: =3 INVITE
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, =PTIONS, =YE, REFER
  Contact: =lt;sip:asterisk at xxx.xxx.xxx.xxx>
  Content-Type: =pplication/sdp
  Content-Length: 219

  =P>v=0
  o=root 17002 17015 IN IP4 =xx.xxx.xxx.xxx
  s=session
  c=IN IP4 195.216.65.216
  t=0 =
  m=audio =4276 RTP/AVP 3 101
  a=rtpmap:3 GSM/8000
  a=rtpmap:101 =elephone-event/8000
  a=fmtp:101 0-16
  a=silenceSupp:off - - - =

  =P> to =xx.xxx.xxx.xxx:2841

  =P> 

  =P>*********************** NOT WORKING =*******************************

  =P>Sip read:
  INVITE sip:4803 at 192.168.1.20 =IP/2.0
  l: 214
  m: <sip:4803 at 192.168.1.111:5060>
  i: 0b367539445=3aa4681709356c705397 at 192.168.1.20
  c: =pplication/sdp
  Max-Forwards: 70
  CSeq: 1 INVITE
  f: =lt;sip:4803 at 192.168.1.111:5060>;tag=41308811925234
  t: =lt;sip:4803 at 192.168.1.20>;tag=as463b04a6
  User-Agent: =JLabs-SJphone/1.30.252
  v: SIP/2.0/UDP =92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227c5f0000648200000=c8

  =P>v=0
  o=- 3318545106 3318545107 IN =P4 =92.168.1.111
  s=SJphone
  c=IN IP4 0.0.0.0
  t=0 =
  a=direction:active
  m=audio 16400 RTP/AVP 3 =01
  a=rtpmap:3 =SM/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 =-11,16

  =P>11 headers, 10 lines
  Using latest =equest as basis =equest
  Sending to 192.168.1.111 : 5060 (NAT)
  We're at =92.168.1.20 port =8336
  Answering/Requesting with root capability =x4 =ulaw)
  Answering with preferred capability 0x2 (gsm)
  Answering =ith =on-codec capability 0x1 (telephone-event)
  Reliably =ransmitting =NAT):
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP =92.168.1.111;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000000c8;re=eived=192.168.1.111;rport=5060
  From: =lt;sip:4803 at 192.168.1.111:5060>;tag=41308811925234
  To: =lt;sip:4803 at 192.168.1.20>;tag=as463b04a6
  Call-ID: 0b367539445=3aa4681709356c705397 at 192.168.1.20
  CSeq: = INVITE
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, =PTIONS, =YE, REFER
  Contact: <sip:4803 at 192.168.1.20>
  Content-Type: =pplication/sdp
  Content-Length: 241

  =P>v=0
  o=root 12791 12793 IN IP4 =92.168.1.111
  s=session
  c=IN IP4 192.168.1.111
  t=0 =
  m=audio 16398 =TP/AVP 0 3 101
  a=rtpmap:0 PCMU/8000
  a=rtpmap:3 =SM/8000
  a=rtpmap:101 =elephone-event/8000
  a=fmtp:101 0-16
  a=silenceSupp:off - - - =

  =P> to 192.168.1.111:5060

  =P>Thanks

  =P>KF

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