[Asterisk-Users] music on hold trouble

Sun Feb 27 19:46:04 MST 2005

Hi All

I seem to have a small problem with the music on hold button on SJPhone.

I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS.

On the rapid dist when I press the music on hold button on my SJPhone I get music on hold.

When I do the same I get no music on hold just silence.
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.

Can anyone help with this?
 is this a bug in CVS?

here are debugs from both installs (1 working and 1 not working):

**********************  WORKING ************************
Sip read:
INVITE sip:asterisk at xxx.xxx.xxx.xxx SIP/2.0
l: 214
m: <sip:4802 at 192.168.1.111:5060>
i: 08c50c2469d676562285f02f72e5f6be at xxx.xxx.xxx.xxx
c: application/sdp
Max-Forwards: 70
CSeq: 13 INVITE
f: <sip:4802 at xxx.xxx.xxx.xxx:2841>;tag=41280171719448
t: <sip:asterisk at xxx.xxx.xxx.xxx>;tag=as7cf27066
User-Agent: SJLabs-SJphone/1.30.252
v: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000078

v=0
o=- 3318544820 3318544833 IN IP4 192.168.1.111
s=SJphone
c=IN IP4 0.0.0.0
t=0 0
a=direction:active
m=audio 16394 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

11 headers, 10 lines
Using latest request as basis request
Sending to 192.168.1.111 : 5060 (NAT)
Found audio format UNKN
Found audio format UNKN
Found description format GSM
Found description format telephone-event
Capabilities: us - 6, them - 2/0, combined - 2
Non-codec capabilities: us - 1, them - 1, combined - 1
We're at xxx.xxx.xxx.xxx port 14276
Answering with preferred capability 2
Answering with non-codec capability 1
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000078;received=xxx.xxx.xxx.xxx
From: <sip:4802 at xxx.xxx.xxx.xxx:2841>;tag=41280171719448
To: <sip:asterisk at xxx.xxx.xxx.xxx>;tag=as7cf27066
Call-ID: 08c50c2469d676562285f02f72e5f6be at xxx.xxx.xxx.xxx
CSeq: 13 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:asterisk at xxx.xxx.xxx.xxx>
Content-Type: application/sdp
Content-Length: 219

v=0
o=root 17002 17015 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 195.216.65.216
t=0 0
m=audio 14276 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to xxx.xxx.xxx.xxx:2841

*********************** NOT WORKING ********************************

Sip read:
INVITE sip:4803 at 192.168.1.20 SIP/2.0
l: 214
m: <sip:4803 at 192.168.1.111:5060>
i: 0b36753944573aa4681709356c705397 at 192.168.1.20
c: application/sdp
Max-Forwards: 70
CSeq: 1 INVITE
f: <sip:4803 at 192.168.1.111:5060>;tag=41308811925234
t: <sip:4803 at 192.168.1.20>;tag=as463b04a6
User-Agent: SJLabs-SJphone/1.30.252
v: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000000c8

v=0
o=- 3318545106 3318545107 IN IP4 192.168.1.111
s=SJphone
c=IN IP4 0.0.0.0
t=0 0
a=direction:active
m=audio 16400 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

11 headers, 10 lines
Using latest request as basis request
Sending to 192.168.1.111 : 5060 (NAT)
We're at 192.168.1.20 port 18336
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000000c8;received=192.168.1.111;rport=5060
From: <sip:4803 at 192.168.1.111:5060>;tag=41308811925234
To: <sip:4803 at 192.168.1.20>;tag=as463b04a6
Call-ID: 0b36753944573aa4681709356c705397 at 192.168.1.20
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4803 at 192.168.1.20>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 12791 12793 IN IP4 192.168.1.111
s=session
c=IN IP4 192.168.1.111
t=0 0
m=audio 16398 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 192.168.1.111:5060

Thanks

KF

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