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@xxx.xxx.xxx.xxx SIP/2.0 l: 214 m: <sip:4802@192.168.1.111:5060> i: 08c50c2469d676562285f02f72e5f6be@xxx.xxx.xxx.xxx c: application/sdp Max-Forwards: 70 CSeq: 13 INVITE f: <sip:4802@xxx.xxx.xxx.xxx:2841>;tag=41280171719448 t: <sip:asterisk@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@xxx.xxx.xxx.xxx:2841>;tag=41280171719448 To: <sip:asterisk@xxx.xxx.xxx.xxx>;tag=as7cf27066 Call-ID: 08c50c2469d676562285f02f72e5f6be@xxx.xxx.xxx.xxx CSeq: 13 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:asterisk@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@192.168.1.20 SIP/2.0 l: 214 m: <sip:4803@192.168.1.111:5060> i: 0b36753944573aa4681709356c705397@192.168.1.20 c: application/sdp Max-Forwards: 70 CSeq: 1 INVITE f: <sip:4803@192.168.1.111:5060>;tag=41308811925234 t: <sip:4803@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@192.168.1.111:5060>;tag=41308811925234 To: <sip:4803@192.168.1.20>;tag=as463b04a6 Call-ID: 0b36753944573aa4681709356c705397@192.168.1.20 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:4803@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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050227/2b6f138d/attachment.htm
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@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@xxx.xxx.xxx.xxx =IP/2.0
l: 214
m: <sip:4802@192.168.1.111:5060>
i: 08c50c24=9d676562285f02f72e5f6be@xxx.xxx.xxx.xxx
c: =pplication/sdp
Max-Forwards: 70
CSeq: 13 INVITE
f: =lt;sip:4802@xxx.xxx.xxx.xxx:2841>;tag=41280171719448
t: =lt;sip:asterisk@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@xxx.xxx.xxx.xxx:2841>;tag=41280171719448
To: =lt;sip:asterisk@xxx.xxx.xxx.xxx>;tag=as7cf27066
Call-ID: 08c50c24=9d676562285f02f72e5f6be@xxx.xxx.xxx.xxx
CSeq: =3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, =PTIONS, =YE, REFER
Contact: =lt;sip:asterisk@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@192.168.1.20 =IP/2.0
l: 214
m: <sip:4803@192.168.1.111:5060>
i: 0b367539445=3aa4681709356c705397@192.168.1.20
c: =pplication/sdp
Max-Forwards: 70
CSeq: 1 INVITE
f: =lt;sip:4803@192.168.1.111:5060>;tag=41308811925234
t: =lt;sip:4803@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@192.168.1.111:5060>;tag=41308811925234
To: =lt;sip:4803@192.168.1.20>;tag=as463b04a6
Call-ID: 0b367539445=3aa4681709356c705397@192.168.1.20
CSeq: = INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, =PTIONS, =YE, REFER
Contact: <sip:4803@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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>I too am having the same problem with =VS from last night. From my >debugging, * never attempts to start MOH. Anyone else =ound this?Me too Music on hold - with SIP handsets at least - stopped working for me with asterisk 1.0.6 and cvs. If I downgraded to 1.0.5 works fine, upgrade and it stops working. all versions work fine if a dial an extension for music on hold. Cheers Walt _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
w fm3 schrieb:>> I too am having the same problem with =VS from last night. From my >> debugging, * never attempts to start MOH. Anyone else =ound this? > > > Me too > > Music on hold - with SIP handsets at least - stopped working for me with > asterisk 1.0.6 and cvs. > > If I downgraded to 1.0.5 works fine, upgrade and it stops working. > > all versions work fine if a dial an extension for music on hold. > > Cheers > > WaltIs there a solution for this Problem, I still have the same! Regards Bastian