[Asterisk-Users] music on hold trouble
Rod Bacon
rod.bacon at empoweredcomms.com.au
Tue Mar 1 19:47:28 MST 2005
Thanks for not flaming me for asking (or at least agreeing with) such a misguided question.
I now realise that playing MOH by pressing the HOLD button on a handset (or via a softphone) is actually a function of the phone itselt, and not *.
I setup an extension to play MOH, and pointed my SIPURA and SNOM phones to it as their "MOH server", and everything works sweetly (notwithstanding a bug in the SNOM firmware).
----- Original Message -----
From: Rod Bacon
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, March 01, 2005 2:21 PM
Subject: Re: [Asterisk-Users] music on hold trouble
I too am having the same problem with =VS from last night. From my debugging, * never attempts to start MOH. Anyone else =ound this?
----- Original Message -----
From: Krystian Filiks
To: asterisk-users at lists.digi=m.com
Sent: Monday, February 28, 2005 =:46 PM
Subject: [Asterisk-Users] music =n hold trouble
Hi =ll
=DIV>
=DIV>I seem to have a =mall problem with the =usic on =old button on SJPhone.
=DIV>
=DIV>I have 2 asterisk =nstallations one =rom the Rapid =istribution and one from the latest CVS.
=DIV>
=DIV>On the rapid dist =hen I press the =usic on hold =utton on my SJPhone I get music on hold.
=DIV>
=DIV>When I do the same I =et no music on =old just =ilence.
=DIV>I create extension =ike this exten =3D> =111,1,MusicOnHold(Default), and when I dial it then I =ear music, so =usic on =old works but the hold button do not.
=DIV>
=DIV>Can anyone help with =his?
=DIV> is this a bug =n CVS?
=DIV>
=DIV>
=DIV>here are debugs from =oth 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: 08c5=c24=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 =3D92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d000=0=78
=P>v=0
o=- =318544820 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 =3D01
a=rtpmap:3 =SM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 =3D-11,16
=P>11 headers, 10 =ines
Using latest =equest as basis =equest
Sending to 192.168.1.111 : 5060 =NAT)
Found audio format =NKN
Found audio format UNKN
Found description =ormat GSM
Found =escription format telephone-event
Capabilities: us = 6, them - 2/0, =ombined = 2
Non-codec capabilities: us - 1, them - 1, =ombined - 1
We're =t xxx.xxx.xxx.xxx port 14276
Answering with =referred =apability =
Answering with non-codec capability 1
Reliably =ransmitting =NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP =3D92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d000=0=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: 08c5=c24=9d676562285f02f72e5f6be at xxx.xxx.xxx.xxx
CSeq: =3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, =3DPTIONS, =YE, REFER
Contact: =3Dlt;sip:asterisk at xxx.xxx.xxx.xxx>
Content-Type: =pplication/sdp
Content-Length: 219
=P>v=0
o=root 17002 =7015 IN IP4 =xx.xxx.xxx.xxx
s=session
c=IN IP4 195.216.65.216
t=0 =3D
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 =3Dxx.xxx.xxx.xxx:2841
=P>
=P>*********************** NOT =ORKING =*******************************
=P>Sip read:
INVITE =ip:4803 at 192.168.1.20 =IP/2.0
l: 214
m: <sip:4803 at 192.168.1.111:5060>
i: 0b36753=445=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 =3D92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000=0=c8
=P>v=0
o=- 3318545106 =318545107 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 =3D01
a=rtpmap:3 =SM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 =3D-11,16
=P>11 headers, 10 =ines
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 =3DNAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP =3D92.168.1.111;branch=z9hG4bKc0a8016f0131c9b142227c5f00006482000000c8;=e=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: 0b36753=445=3aa4681709356c705397 at 192.168.1.20
CSeq: = INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, =3DPTIONS, =YE, REFER
Contact: =lt;sip:4803 at 192.168.1.20>
Content-Type: =pplication/sdp
Content-Length: 241
=P>v=0
o=root 12791 =2793 IN IP4 =92.168.1.111
s=session
c=IN IP4 192.168.1.111
t=0 =3D
m=audio 16398 =TP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 =3DSM/8000
a=rtpmap:101 =elephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - = =
=P> to =92.168.1.111:5060
=P>Thanks
=P>KF
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=P>
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