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 -----
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 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:[EMAIL PROTECTED] =IP/2.0
l: 214
m: <sip:[EMAIL PROTECTED]:5060>
i: [EMAIL PROTECTED]
c: =pplication/sdp
Max-Forwards: 70
CSeq: 13 INVITE
f: =lt;sip:[EMAIL PROTECTED]:2841>;tag=41280171719448
t: =lt;sip:[EMAIL PROTECTED]>;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:[EMAIL PROTECTED]:2841>;tag=41280171719448
To: =lt;sip:[EMAIL PROTECTED]>;tag=as7cf27066
Call-ID: [EMAIL PROTECTED]
CSeq: =3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, =PTIONS, =YE, REFER
Contact: =lt;sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED] =IP/2.0
l: 214
m: <sip:[EMAIL PROTECTED]:5060>
i: [EMAIL PROTECTED]
c: =pplication/sdp
Max-Forwards: 70
CSeq: 1 INVITE
f: =lt;sip:[EMAIL PROTECTED]:5060>;tag=41308811925234
t: =lt;sip:[EMAIL PROTECTED]>;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:[EMAIL PROTECTED]:5060>;tag=41308811925234
To: =lt;sip:[EMAIL PROTECTED]>;tag=as463b04a6
Call-ID: [EMAIL PROTECTED]
CSeq: = INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, =PTIONS, =YE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-user
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to