Annoying hold music

Asterisk Asterisk Support
1

I have a new Asterisk 1.6 server that interfaces with our existing Mitel 3300. Outbound calls from Asterisk are handed off to Mitel, and I’m getting an odd behavior. When I dial out via an Asterisk extension, I’m put on hold and I don’t hear ringing, but rather the Asterisk hold music. When the line answers, the hold music stops and the call proceeds as normal, so it’s not a huge issue. However, I know that any users I put on this system are going to be mystified by this behavior. Here is a typical call log:

-- Executing [XXXXXXX@domain.org:1] Dial("SIP/203-fc006ad8", "SIP/9XXXXXXX@3300") in new stack

== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
– Called 9XXXXXXX@3300
– Call on SIP/3300-98001758 placed on hold
– Started music on hold, class ‘default’, on SIP/203-fc006ad8
– SIP/3300-98001758 is making progress passing it to SIP/203-fc006ad8

I find the “placed on hold” entry interesting. Is Asterisk doing that or is my 3300? Furthermore, this oddly does not happen roughly 10% of the time. In 1 out of every 10 calls or so, I get no hold music and can hear the ringing of the outbound call.

Any ideas?

2

The 3300, which you will be able to confirm by looking for the incoming re-invite in the sip set debug or sip history output for the call.

As this is step 1, it looks like the Mitel is also answering the call prematurely, thus taking responsibility for any ringback tone generation. It is just possible this is call progress, rather than answer, in which case it might be worth trying an “r” option on the Dial. However that will prevent the caller hearing any real call progress message.

3

Hi

need a sip debug of this as its not what ive seen before with Asterisk and Mitel integration.

Ian

4

Here’s the SIP debug. The external number is “8235234”, the Asterisk extension is 203 (192.168.2.159). Asterisk is served from 192.168.1.99 and the Mitel is 172.16.1.2.

INVITE sip:8235234@sip.domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-4ec83cdadf0af98f-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.2.159:1684
To: sip:8235234@sip.domain.com
From: "jstrope"sip:203@sip.domain.com;tag=970dd060
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.5.4 stamp 53962
Content-Length: 322

v=0
o=- 2 2 IN IP4 192.168.2.159
s=CounterPath Bria Professional
c=IN IP4 192.168.2.159
t=0 0
m=audio 21146 RTP/AVP 107 0 8 18 101
a=sendrecv
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : pVBljPk2 ajppurr/ 192.168.2.159 21146

<------------->
— (13 headers 13 lines) —
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
Sending to 192.168.2.159 : 1684 (NAT)
Using INVITE request as basis request - M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
Found peer ‘203’ for ‘203’ from 192.168.2.159:1684

<— Reliably Transmitting (NAT) to 192.168.2.159:1684 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-4ec83cdadf0af98f-1—d8754z-;received=192.168.2.159;rport=1684
From: "jstrope"sip:203@sip.domain.com;tag=970dd060
To: sip:8235234@sip.domain.com;tag=as27f5dbd0
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="21b3dd35"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.’ in 6400 ms (Method: INVITE)
social*CLI>
<— SIP read from UDP://192.168.2.159:1684 —>
ACK sip:8235234@sip.domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-4ec83cdadf0af98f-1—d8754z-;rport
Max-Forwards: 70
To: sip:8235234@sip.domain.com;tag=as27f5dbd0
From: "jstrope"sip:203@sip.domain.com;tag=970dd060
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
social*CLI>
<— SIP read from UDP://192.168.2.159:1684 —>
INVITE sip:8235234@sip.domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-9b8390d5482570b6-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.2.159:1684
To: sip:8235234@sip.domain.com
From: “jstrope"sip:203@sip.domain.com;tag=970dd060
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.5.4 stamp 53962
Authorization: Digest username=“203”,realm=“asterisk”,nonce=“21b3dd35”,uri="sip:8235234@sip.domain.com”,response=“b9449758be3e24b6fa4829982a0b6223”,algorithm=MD5
Content-Length: 322

v=0
o=- 2 2 IN IP4 192.168.2.159
s=CounterPath Bria Professional
c=IN IP4 192.168.2.159
t=0 0
m=audio 21146 RTP/AVP 107 0 8 18 101
a=sendrecv
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : pVBljPk2 ajppurr/ 192.168.2.159 21146

<------------->
— (14 headers 13 lines) —
Sending to 192.168.2.159 : 1684 (NAT)
Using INVITE request as basis request - M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
Found peer ‘203’ for ‘203’ from 192.168.2.159:1684
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.159:21146
Found unknown media description format BV32 for ID 107
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10050f (g723|gsm|ulaw|alaw|g729|ilbc|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.2.159:21146
Peer video RTP is at port 192.168.2.159:252
Looking for 8235234 in intrahealth.org (domain sip.domain.com)
list_route: hop: sip:203@192.168.2.159:1684

<— Transmitting (NAT) to 192.168.2.159:1684 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-9b8390d5482570b6-1—d8754z-;received=192.168.2.159;rport=1684
From: "jstrope"sip:203@sip.domain.com;tag=970dd060
To: sip:8235234@sip.domain.com
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:8235234@192.168.1.99
Content-Length: 0

<------------>
– Executing [8235234@intrahealth.org:1] Dial(“SIP/203-fc005598”, “SIP/98235234@3300”) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
Audio is at 192.168.1.99 port 13252
Video is at 192.168.1.99 port 19526
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.16.1.2:5060:
INVITE sip:98235234@172.16.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK44dcbee6;rport
Max-Forwards: 70
From: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
To: sip:98235234@172.16.1.2
Contact: sip:9193136203@192.168.1.99
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Mon, 14 Sep 2009 13:58:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 376

v=0
o=root 1959301361 1959301361 IN IP4 192.168.1.99
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.1.99
b=CT:384
t=0 0
m=audio 13252 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

-- Called 98235234@3300

social*CLI>
<— SIP read from UDP://172.16.1.2:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.99:5060;received=172.16.1.1;branch=z9hG4bK44dcbee6;rport
From: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
To: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
social*CLI>
<— SIP read from UDP://172.16.1.2:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.99:5060;received=172.16.1.1;branch=z9hG4bK44dcbee6;rport
From: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
To: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 102 INVITE
Contact: sip:98235234@172.16.1.2:5060;transport=udp
Content-Type: application/sdp
Content-Length: 201

v=0
o=- 1714 1714 IN IP4 0.0.0.0
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 50052 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-mitel-dtmf-det-required:yes
m=video 0 RTP/AVP 34

<------------->
— (9 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found RTP video format 34
Peer audio RTP is at port 0.0.0.0:50052
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80004 (ulaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 0.0.0.0:50052
– Call on SIP/3300-98001758 placed on hold
Audio is at 192.168.1.99 port 11682
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 192.168.2.159:1684 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-9b8390d5482570b6-1—d8754z-;received=192.168.2.159;rport=1684
From: "jstrope"sip:203@sip.domain.com;tag=970dd060
To: sip:8235234@sip.domain.com;tag=as269236e9
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:8235234@192.168.1.99
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 815554457 815554457 IN IP4 192.168.1.99
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.1.99
t=0 0
m=audio 11682 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Started music on hold, class ‘default’, on SIP/203-fc005598
– SIP/3300-98001758 is making progress passing it to SIP/203-fc005598
Reliably Transmitting (NAT) to 192.168.1.131:5060:
OPTIONS sip:334@192.168.1.131;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK3578601c;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.99;tag=as5e45a867
To: sip:334@192.168.1.131;transport=udp
Contact: sip:asterisk@192.168.1.99
Call-ID: 35b0704c551aeff47bef88b5524b08f3@192.168.1.99
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.1.1
Date: Mon, 14 Sep 2009 13:58:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

social*CLI>
<— SIP read from UDP://192.168.1.131:5060 —>
SIP/2.0 200 OK
Call-ID: 35b0704c551aeff47bef88b5524b08f3@192.168.1.99
CSeq: 102 OPTIONS
From: “asterisk” sip:asterisk@192.168.1.99;tag=as5e45a867
To: sip:334@192.168.1.131;tag=92b9644f0fe20a0
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK3578601c;rport
Content-Length: 0
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Contact: sip:334@192.168.1.131;transport=udp
Supported: replaces
User-Agent: Aastra/TalkSwitch 9112i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘35b0704c551aeff47bef88b5524b08f3@192.168.1.99’ Method: OPTIONS
social*CLI>
<— SIP read from UDP://172.16.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.99:5060;received=172.16.1.1;branch=z9hG4bK44dcbee6;rport
From: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
To: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 102 INVITE
Supported: timer
Contact: sip:98235234@172.16.1.2:5060;transport=udp
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
Content-Type: application/sdp
User-Agent: Mitel-3300-ICP 9.0.0.41
Content-Length: 201

v=0
o=- 1714 1714 IN IP4 0.0.0.0
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 50052 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-mitel-dtmf-det-required:yes
m=video 0 RTP/AVP 34

<------------->
— (12 headers 10 lines) —
list_route: hop: sip:98235234@172.16.1.2:5060;transport=udp
set_destination: Parsing sip:98235234@172.16.1.2:5060;transport=udp for address/port to send to
set_destination: set destination to 172.16.1.2, port 5060
Transmitting (NAT) to 172.16.1.2:5060:
ACK sip:98235234@172.16.1.2:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK2c5d0190;rport
Max-Forwards: 70
From: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
To: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
Contact: sip:9193136203@192.168.1.99
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0

-- SIP/3300-98001758 answered SIP/203-fc005598
-- Stopped music on hold on SIP/203-fc005598

Audio is at 192.168.1.99 port 11682
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.2.159:1684 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-9b8390d5482570b6-1—d8754z-;received=192.168.2.159;rport=1684
From: "jstrope"sip:203@sip.domain.com;tag=970dd060
To: sip:8235234@sip.domain.com;tag=as269236e9
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:8235234@192.168.1.99
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 815554457 815554458 IN IP4 192.168.1.99
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.1.99
t=0 0
m=audio 11682 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
social*CLI>
<— SIP read from UDP://192.168.2.159:1684 —>
ACK sip:8235234@192.168.1.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-f95028baabcc49c1-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.2.159:1684
To: sip:8235234@sip.domain.com;tag=as269236e9
From: “jstrope"sip:203@sip.domain.com;tag=970dd060
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 2 ACK
User-Agent: Bria Professional release 2.5.4 stamp 53962
Authorization: Digest username=“203”,realm=“asterisk”,nonce=“21b3dd35”,uri="sip:8235234@sip.domain.com”,response=“b9449758be3e24b6fa4829982a0b6223”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
social*CLI>
<— SIP read from UDP://172.16.1.2:5060 —>
INVITE sip:9193136203@192.168.1.99 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.2:5060;branch=z9hG4bK2858895472-61462060
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
Supported: timer,replaces
From: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
To: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 2 INVITE
Contact: sip:98235234@172.16.1.2:5060;transport=udp
Content-Type: application/sdp
Content-Length: 207

v=0
o=- 1714 1715 IN IP4 172.16.1.3
s=-
c=IN IP4 172.16.1.3
t=0 0
m=audio 50052 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-mitel-dtmf-det-required:yes
m=video 0 RTP/AVP 34

<------------->
— (12 headers 10 lines) —
Sending to 172.16.1.2 : 5060 (NAT)
Found RTP audio format 0
Found RTP audio format 101
Found RTP video format 34
Peer audio RTP is at port 172.16.1.3:50052
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80004 (ulaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.1.3:50052

<— Transmitting (NAT) to 172.16.1.2:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.2:5060;branch=z9hG4bK2858895472-61462060;received=172.16.1.2
From: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
To: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:9193136203@192.168.1.99
Content-Length: 0

<------------>
Audio is at 192.168.1.99 port 13252
Video is at 192.168.1.99 port 19526
Adding codec 0x4 (ulaw) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 172.16.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.1.2:5060;branch=z9hG4bK2858895472-61462060;received=172.16.1.2
From: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
To: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:9193136203@192.168.1.99
Content-Type: application/sdp
Content-Length: 329

v=0
o=root 1959301361 1959301362 IN IP4 192.168.1.99
s=Asterisk PBX 1.6.1.1
c=IN IP4 192.168.1.99
b=CT:384
t=0 0
m=audio 13252 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

<------------>
social*CLI>
<— SIP read from UDP://172.16.1.2:5060 —>
ACK sip:9193136203@192.168.1.99 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.2:5060;branch=z9hG4bK2858945472-61462061
Max-Forwards: 70
From: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
To: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 2 ACK
Contact: sip:98235234@172.16.1.2:5060;transport=udp
Content-Length: 0

<------------->
— (9 headers 0 lines) —
social*CLI>
<— SIP read from UDP://192.168.2.159:1684 —>

<------------->
social*CLI>
<— SIP read from UDP://192.168.2.159:1684 —>
BYE sip:8235234@192.168.1.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-cd9db5fd8a5d3659-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.2.159:1684
To: sip:8235234@sip.domain.com;tag=as269236e9
From: “jstrope"sip:203@sip.domain.com;tag=970dd060
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 3 BYE
User-Agent: Bria Professional release 2.5.4 stamp 53962
Authorization: Digest username=“203”,realm=“asterisk”,nonce=“21b3dd35”,uri="sip:8235234@192.168.1.99”,response=“197a41cc52804330ba765a6ac750e12a”,algorithm=MD5
Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.2.159 : 1684 (NAT)

<— Transmitting (NAT) to 192.168.2.159:1684 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.159:1684;branch=z9hG4bK-d8754z-cd9db5fd8a5d3659-1—d8754z-;received=192.168.2.159;rport=1684
From: "jstrope"sip:203@sip.domain.com;tag=970dd060
To: sip:8235234@sip.domain.com;tag=as269236e9
Call-ID: M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.
CSeq: 3 BYE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘76324a65395c089e2ac839e20515b929@192.168.1.99’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:98235234@172.16.1.2:5060;transport=udp for address/port to send to
set_destination: set destination to 172.16.1.2, port 5060
Reliably Transmitting (NAT) to 172.16.1.2:5060:
BYE sip:98235234@172.16.1.2:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK33c46c0d;rport
Max-Forwards: 70
From: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
To: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.1.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (intrahealth.org, 8235234, 1) exited non-zero on 'SIP/203-fc005598’
social*CLI>
<— SIP read from UDP://172.16.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK33c46c0d;rport
From: “jstrope” sip:9193136203@192.168.1.99;tag=as17580e47
To: sip:98235234@172.16.1.2;tag=0_2846575472-61462059
Call-ID: 76324a65395c089e2ac839e20515b929@192.168.1.99
CSeq: 103 BYE
Contact: sip:98235234@172.16.1.2:5060;transport=udp
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
User-Agent: Mitel-3300-ICP 9.0.0.41
Content-Length: 0

<------------->
— (10 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘76324a65395c089e2ac839e20515b929@192.168.1.99’ Method: ACK
Really destroying SIP dialog ‘M2NjMDFjZjFlN2ZjNzRjN2JhMWUyNmE5MzFhODcwZjM.’ Method: BYE

5

The Mitel has responded with PROGRESS, which has forced Asterisk to pass audio, and hold (the 0.0.0.0 addresses in the SDP), and means that the Mitel has take responsibility for generating tones.

SIP/2.0 183 Session Progress
Content-Type: application/sdp
…

v=0
o=- 1713 1713 IN IP4 0.0.0.0
s=-
c=IN IP4 0.0.0.0
…

6

Hmm…thanks for the response. Any way around this on the Asterisk side?

7

As I said, try “r” as a Dial option, but you may lose call progress indications that you really want.

8

One way or another, you could always configure silent MOH for the call.

9

The “r” option gives me one ring (instead of none) then quickly gives me the hold music. I think I’ll actually create some hold music that simply sounds like a ring…

10

Hmm…I’m unable to get rid of this hold music without just doing an unload on the music on hold module. The dialplan here is pretty simple:

exten => _XXXXXXX,1,Dial(SIP/9${EXTEN}@3300)

I’ve tried adding a macro that turns off hold music via SetMusicOnHold(none) and Set(CHANNEL(musicclass)=none) via a macro executed in the dial plan. Neither works.

This is my musiconhold.conf file:

[general]

[none]
mode=files
directory=/dev/null

[default]
mode=files
directory=/var/lib/asterisk/moh

What is the correct syntax to add to the dialing plan to disable music on hold for the calling party?