Asterisk 1.8.x AAstra 9133i, 9143i, 480i NO MUSIC ON HOLD


#1

Was previously using Asterisk 1.6.X versions , company switched out servers I installed 1.8.X ( tried multiple version ) and when we place a call on hold on any aastra phone in the business 9133i, 9143i, 480i It will not play music on hold… Our Polycom phones work fine, our PAP2T work fine… Anyone have any idea what might be going on I have talked to multiple distributors, aastra , been asking on IRC for over a week
PLEASE HELP !!!


#2

I am having this same problem with 1.8 - the Aastra 9133i phones do not seem to let asterisk know that a call is on hold, even though our Aastra 57i’s do. Very frustrating.

Has anyone managed to find a solution to this problem?


#3

You need to provide sip set debug output starting from the point where you try it initiate the hold. You would normally expect a re-invite with SDP containing “a=sendonly”.


#4

Ok… this is what I get from my 9133i when I turn on sip debug and place a call on hold. Does it help?

<— SIP read from UDP:10.44.13.22:5060 —>
INVITE sip:07814378082@10.44.13.71:5060 SIP/2.0
Via: SIP/2.0/UDP 10.44.13.22:5060;branch=z9hG4bK2ef2649d2
Max-Forwards: 70
Content-Length: 221
To: “07814378082” sip:07814378082@10.44.13.71;tag=as4f7b71d7
From: sip:1314@10.44.13.22:5060;tag=9a877aa7e43d50d
Call-ID: 3bf2947f0681240b1860d2df01816fd6@10.44.13.71:5060
CSeq: 1585270979 INVITE
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Content-Type: application/sdp
Contact: Paul Caligari sip:1314@10.44.13.22:5060
Supported: replaces
User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

v=0
o=MxSIP 0 1794587700 IN IP4 10.44.13.22
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 3000 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
— (15 headers 11 lines) —
Sending to 10.44.13.22:5060 (no NAT)
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 0.0.0.0:3000

<— Transmitting (no NAT) to 10.44.13.22:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.44.13.22:5060;branch=z9hG4bK2ef2649d2;received=10.44.13.22
From: sip:1314@10.44.13.22:5060;tag=9a877aa7e43d50d
To: “07814378082” sip:07814378082@10.44.13.71;tag=as4f7b71d7
Call-ID: 3bf2947f0681240b1860d2df01816fd6@10.44.13.71:5060
CSeq: 1585270979 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:07814378082@10.44.13.71:5060
Content-Length: 0

<------------>
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (no NAT) to 10.44.13.22:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.44.13.22:5060;branch=z9hG4bK2ef2649d2;received=10.44.13.22
From: sip:1314@10.44.13.22:5060;tag=9a877aa7e43d50d
To: “07814378082” sip:07814378082@10.44.13.71;tag=as4f7b71d7
Call-ID: 3bf2947f0681240b1860d2df01816fd6@10.44.13.71:5060
CSeq: 1585270979 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:07814378082@10.44.13.71:5060
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1502122030 1502122031 IN IP4 10.44.13.71
s=Asterisk PBX 1.8.6.0
c=IN IP4 10.44.13.71
t=0 0
m=audio 11756 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (no NAT) to 10.44.13.22:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.44.13.22:5060;branch=z9hG4bK2ef2649d2;received=10.44.13.22
From: sip:1314@10.44.13.22:5060;tag=9a877aa7e43d50d
To: “07814378082” sip:07814378082@10.44.13.71;tag=as4f7b71d7
Call-ID: 3bf2947f0681240b1860d2df01816fd6@10.44.13.71:5060
CSeq: 1585270979 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:07814378082@10.44.13.71:5060
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1502122030 1502122031 IN IP4 10.44.13.71
s=Asterisk PBX 1.8.6.0
c=IN IP4 10.44.13.71
t=0 0
m=audio 11756 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:10.44.13.22:5060 —>
ACK sip:07814378082@10.44.13.71:5060 SIP/2.0
Via: SIP/2.0/UDP 10.44.13.22:5060;branch=z9hG4bK2d3d54881
Max-Forwards: 70
Content-Length: 0
To: “07814378082” sip:07814378082@10.44.13.71;tag=as4f7b71d7
From: sip:1314@10.44.13.22:5060;tag=9a877aa7e43d50d
Call-ID: 3bf2947f0681240b1860d2df01816fd6@10.44.13.71:5060
CSeq: 1585270979 ACK
Contact: Paul Caligari sip:1314@10.44.13.22:5060
User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:10.44.13.22:5060 —>
ACK sip:07814378082@10.44.13.71:5060 SIP/2.0
Via: SIP/2.0/UDP 10.44.13.22:5060;branch=z9hG4bK2d3d54881
Max-Forwards: 70
Content-Length: 0
To: “07814378082” sip:07814378082@10.44.13.71;tag=as4f7b71d7
From: sip:1314@10.44.13.22:5060;tag=9a877aa7e43d50d
Call-ID: 3bf2947f0681240b1860d2df01816fd6@10.44.13.71:5060
CSeq: 1585270979 ACK
Contact: Paul Caligari sip:1314@10.44.13.22:5060
User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45


#5

It’s using the obsolete method of setting the connection address to 0.0.0.0. I do seem to remember that there is/was a problem with that. Search issues.asterisk.org for details.