[asterisk-users] a=sendonly Music On Hold ignored
Michael
voip.question at gmail.com
Thu Jul 21 03:24:00 CDT 2011
On Tue, Jul 19, 2011 at 10:12 PM, Matthew J. Roth <mroth at imminc.com> wrote:
> Michael wrote:
> >
> > True. In the working system, LAN calls are also using G.729, while
> > in the non-working system, LAN calls are in G.711 (supported but
> > not prioritized by the phones) and only the SIP trunk to the ITSP
> > is set to G.729.
>
> Can you set the phone to G.711 and try making a LAN call on the non-
> working system. If a call that is G.711 from end-to-end doesn't have
> the same problem it would be evidence of a codec translation issue.
>
That's the full trace of a call in G.711:
Reliably Transmitting (no NAT) to 192.168.1.109:5060:
INVITE sip:500 at 192.168.1.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4751ec69;rport
Max-Forwards: 70
From: "9000" <sip:9000 at 192.168.1.10>;tag=as40d1788b
To: <sip:500 at 192.168.1.109:5060>
Contact: <sip:9000 at 192.168.1.10>
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.6.2.19)
Date: Thu, 21 Jul 2011 07:12:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 1017783905 1017783905 IN IP4 192.168.1.10
s=Asterisk PBX 1.6.2.19
c=IN IP4 192.168.1.10
t=0 0
m=audio 17696 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called 500
<--- SIP read from UDP:192.168.1.109:5060 --->
SIP/2.0 180 Ringing
From: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
To: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;branch=z9hG4bK4751ec69
Supported: replaces,100rel
User-Agent: SIP Phone
Contact: <sip:500 at 192.168.1.109:5060>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- SIP/500-00000330 is ringing
<--- SIP read from UDP:192.168.1.109:5060 --->
SIP/2.0 200 OK
From: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
To: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;branch=z9hG4bK4751ec69
Supported: replaces,100rel
User-Agent: SIP Phone
Contact: <sip:500 at 192.168.1.109:5060>
Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 226
v=0
o=500 1096000000 0 IN IP4 192.168.1.109
s=Audio Session
i=Audio Session
c=IN IP4 192.168.1.109
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.109:16384
list_route: hop: <sip:500 at 192.168.1.109:5060>
set_destination: Parsing <sip:500 at 192.168.1.109:5060> for address/port to
send to
set_destination: set destination to 192.168.1.109, port 5060
Transmitting (no NAT) to 192.168.1.109:5060:
ACK sip:500 at 192.168.1.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1265e92e;rport
Max-Forwards: 70
From: "9000" <sip:9000 at 192.168.1.10>;tag=as40d1788b
To: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
Contact: <sip:9000 at 192.168.1.10>
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.6.2.19)
Content-Length: 0
---
-- SIP/500-00000330 answered SIP/Smile-0000032f
[Jul 21 10:12:36] WARNING[13622]: dsp.c:1360 ast_dsp_process: Inband DTMF is
not supported on codec g729. Use RFC2833
<--- SIP read from UDP:192.168.1.109:5060 --->
INVITE sip:9000 at 192.168.1.10 SIP/2.0
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-313-c04d6-373f24ae
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SIP Phone
Contact: <sip:500 at 192.168.1.109:5060>
Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 294
v=0
o=500 1096000001 0 IN IP4 192.168.1.109
s=SIPPhone Session
i=Audio Session
c=IN IP4 0.0.0.0
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=sendonly
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (13 headers 14 lines) ---
Sending to 192.168.1.109 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.1.109:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.109:5060
;branch=z9hG4bK-313-c04d6-373f24ae;received=192.168.1.109
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.6.2.19)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:9000 at 192.168.1.10>
Content-Length: 0
<------------>
Audio is at 192.168.1.10 port 17696
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.1.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.109:5060
;branch=z9hG4bK-313-c04d6-373f24ae;received=192.168.1.109
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.6.2.19)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:9000 at 192.168.1.10>
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 1017783905 1017783905 IN IP4 192.168.1.10
s=Asterisk PBX 1.6.2.19
c=IN IP4 192.168.1.10
t=0 0
m=audio 17696 RTP/AVP 0 101
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:192.168.1.109:5060 --->
ACK sip:9000 at 192.168.1.10 SIP/2.0
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-313-c0521-2b8a87aa
Max-Forwards: 70
User-Agent: SIP Phone
Contact: <sip:500 at 192.168.1.109:5060>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.109:5060 --->
INVITE sip:9000 at 192.168.1.10 SIP/2.0
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-321-c3cbc-39b225e8
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SIP Phone
Contact: <sip:500 at 192.168.1.109:5060>
Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 300
v=0
o=500 1096000000 0 IN IP4 192.168.1.109
s=SIPPhone Session
i=Audio Session
c=IN IP4 192.168.1.109
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (13 headers 14 lines) ---
Sending to 192.168.1.109 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.1.109:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.109:5060
;branch=z9hG4bK-321-c3cbc-39b225e8;received=192.168.1.109
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.6.2.19)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:9000 at 192.168.1.10>
Content-Length: 0
<------------>
Audio is at 192.168.1.10 port 17696
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.1.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.109:5060
;branch=z9hG4bK-321-c3cbc-39b225e8;received=192.168.1.109
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.6.2.19)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:9000 at 192.168.1.10>
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 1017783905 1017783905 IN IP4 192.168.1.10
s=Asterisk PBX 1.6.2.19
c=IN IP4 192.168.1.10
t=0 0
m=audio 17696 RTP/AVP 0 101
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:192.168.1.109:5060 --->
ACK sip:9000 at 192.168.1.10 SIP/2.0
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-322-c3ce9-34d41f51
Max-Forwards: 70
User-Agent: SIP Phone
Contact: <sip:500 at 192.168.1.109:5060>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.109:5060 --->
BYE sip:9000 at 192.168.1.10 SIP/2.0
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 3 BYE
Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-32b-c625a-41626b8
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SIP Phone
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.109 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.1.109:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.109:5060
;branch=z9hG4bK-32b-c625a-41626b8;received=192.168.1.109
From: <sip:500 at 192.168.1.109:5060
>;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000"<sip:9000 at 192.168.1.10>;tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c at 192.168.1.10
CSeq: 3 BYE
Server: FPBX-2.8.1(1.6.2.19)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
The same thing happens (or doesn't happen). MOH is not initiated when
sendonly is received by Asterisk. I'm not an expert on this point, but I
suspect that it's a system parameter somewhere and not related to codecs.
>
> > We tested with NAT set to "no" and "yes" and neither settings
> > mattered.
>
> As long as the phone and the Asterisk server are both on the same LAN,
> my recommendation would be to test with NAT set to "no". NAT is not
> necessary unless there is a firewall between the phone and the
> Asterisk server and setting it to "no" also eliminates a variable that
> differentiates it from the working system.
>
We changed it to "no". It doesn't matter.
>
> > It should (have modules loaded for both formats). How do we check
> > this?
>
> The following command should output a line for each module (as shown):
>
> # asterisk -rx 'module show' | egrep 'format_g729.so|format_pcm.so'
> format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 5
> format_g729.so Raw G729 data 0
>
That's what I get:
[root at pbx ~]# asterisk -rx 'module show' | egrep
'format_g729.so|format_pcm.so'
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.
0
format_g729.so Raw G729 data
0
What does the 5 (or in my case 0) stand for?
Thanks.
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