*13.4-SIP- no early media even after 183 Session Progress

re-posting, as I mistakenly posted the same in General Section. thank you david55

Hello,

RTP does not flows-out after I get Progress on the channel, but as soon as the channel is answered RTP starts flowing-out.

I have “Asterisk 13.4.0” and I am trying to playback a prompt without answering the call, but unable to achieve that.

[default]
exten => 99966,1,Goto(noanswer_demo,s,1)

[noanswer_demo]
exten => s,1,Progress()
exten => s,n,Monitor(wav,testCall-${STRFTIME(${EPOCH},GMT+3,%C%y%m%d%H%M)})
exten => s,n,Background(hello-world, n) ; ----> 1
exten => s,n,WaitExten(1)
exten => s,n(cont),Playback(hello-world, noanswer) ; ----> 2
exten => s,n,Wait(1)
exten => s,n,Playback(hello-world) ; ----> 3
exten => s,n,Playback(demo-instruct)
exten => s,n,Playback(hello-world)
exten => s,n,Playback(demo-instruct)
exten => s,n,Hangup()

NOTE: the recorded file (testCall-201508241206.wav in this case) has all the prompts in it (starting from 1,2,3 …)
BUT called-party only hears from 3 onward, after call get answered.

here are the logs of verbose, SIP debug and RTP debug


usman@my-PBX01:~$ sudo asterisk -r
Asterisk 13.4.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.4.0 currently running on my-PBX01 (pid = 13817)


<--- SIP read from UDP:far_end_ip:5060 --->
INVITE tel:99966;phone-context=unknown SIP/2.0
Content-Length:1060
From:<tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To:<tel:99966;phone-context=unknown>
Via:SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bKYDZcf.ZZ307DDajg
Call-ID:712DFF13385DB12F5FD6136D@6443ffffffff
CSeq:1 INVITE
Max-Forwards:70
Route:<sip:my_ast_ip:5060;lr>
Record-Route:<sip:AAQAAZK5DAAAvAAAAvQAAfHoK@far_end_ip:5060;lr>
Request-Disposition:no-fork
P-Asserted-Identity:<sip:+96597834852@kw.zain.com;user=phone>,<tel:+96597834852>
Session-Expires:1800;refresher=uac
Contact:sip:far_end_ip:5060
Supported:100rel,timer
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE
P-Charging-Vector:icid-value=aZWWkCSMvZAYAAAAEKDMBWRD3BA-;icid-generated-at=far_end_ip;orig-ioi=kw.zain.com
P-Early-Media:supported
Content-Type:multipart/mixed;boundary=A0383941CB9FEB5DB7944D9B
MIME-Version:1.0

--A0383941CB9FEB5DB7944D9B
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 far_end_owner_ip
s=-
c=IN IP4 far_end_media_plan_ip
t=0 0
m=audio 8682 RTP/AVP 96 97 98 99 100 101 102 8 103
b=AS:80
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-set=0,2,4,7; mode-change-period=2; mode-change-neighbor=1; max-red=0
a=rtpmap:97 AMR/8000
a=fmtp:97 mode-set=7; max-red=0
a=rtpmap:98 AMR/8000
a=fmtp:98 mode-set=0,2; mode-change-period=2; mode-change-neighbor=1; max-red=0
a=rtpmap:99 AMR/8000
a=fmtp:99 mode-set=4; max-red=0
a=rtpmap:100 AMR/8000
a=fmtp:100 mode-set=2; max-red=0
a=rtpmap:101 AMR/8000
a=fmtp:101 mode-set=0; max-red=0
a=rtpmap:102 GSM-EFR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 telephone-event/8000
a=ptime:20
a=maxptime:20
a=3gOoBTC

--A0383941CB9FEB5DB7944D9B
Content-Type:application/ISUP;base=itu-t92+;version=itu-t92+
Content-Disposition:signal;handling=required
Content-Transfer-Encoding:binary

 
<------------->
--- (20 headers 36 lines) ---
Sending to far_end_ip:5060 (no NAT)
Sending to far_end_ip:5060 (no NAT)
Using INVITE request as basis request - 712DFF13385DB12F5FD6136D@6443ffffffff
No matching peer for '+96597834852' from 'far_end_ip:5060'
  == Using SIP RTP CoS mark 5
Looking for 99966 in default (domain )
sip_route_dump: route/path hop: <sip:AAQAAZK5DAAAvAAAAvQAAfHoK@far_end_ip:5060;lr>

<--- Transmitting (no NAT) to far_end_ip:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bKYDZcf.ZZ307DDajg;received=far_end_ip
Record-Route: <sip:AAQAAZK5DAAAvAAAAvQAAfHoK@far_end_ip:5060;lr>
From: <tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To: <tel:99966;phone-context=unknown>
Call-ID: 712DFF13385DB12F5FD6136D@6443ffffffff
CSeq: 1 INVITE
Server: RVT_DN_PBX_01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:99966@my_ast_ip:5060>
Content-Length: 0


<------------>
    -- Executing [99966@default:1] GotoIf("SIP/+96597834852-00000000", "0?dblookup") in new stack
    -- Executing [99966@default:2] Goto("SIP/+96597834852-00000000", "noanswer_demo,s,1") in new stack
    -- Goto (noanswer_demo,s,1)
    -- Executing [s@noanswer_demo:1] Progress("SIP/+96597834852-00000000", "") in new stack
Audio is at 11646
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec gsm to SDP

<--- Transmitting (no NAT) to far_end_ip:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bKYDZcf.ZZ307DDajg;received=far_end_ip
Record-Route: <sip:AAQAAZK5DAAAvAAAAvQAAfHoK@far_end_ip:5060;lr>
From: <tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To: <tel:99966;phone-context=unknown>;tag=as2dfd8460
Call-ID: 712DFF13385DB12F5FD6136D@6443ffffffff
CSeq: 1 INVITE
Server: RVT_DN_PBX_01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:99966@my_ast_ip:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 230

v=0
o=DN01 1371445892 1371445892 IN IP4 my_media_plan_ip
s=DNIVR01
c=IN IP4 my_media_plan_ip
t=0 0
m=audio 11646 RTP/AVP 8 0 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
    -- Executing [s@noanswer_demo:2] Monitor("SIP/+96597834852-00000000", "wav,testCall-201508241206") in new stack
    -- Executing [s@noanswer_demo:3] BackGround("SIP/+96597834852-00000000", "hello-world, n") in new stack
    -- <SIP/+96597834852-00000000> Playing 'hello-world.gsm' (language 'en')
    -- Executing [s@noanswer_demo:4] WaitExten("SIP/+96597834852-00000000", "1") in new stack
    -- Timeout on SIP/+96597834852-00000000, continuing...
    -- Executing [s@noanswer_demo:5] Playback("SIP/+96597834852-00000000", "hello-world, noanswer") in new stack
    -- <SIP/+96597834852-00000000> Playing 'hello-world.gsm' (language 'en')
    -- Executing [s@noanswer_demo:6] Wait("SIP/+96597834852-00000000", "1") in new stack
    -- Executing [s@noanswer_demo:7] Playback("SIP/+96597834852-00000000", "hello-world") in new stack
Audio is at 11646
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec gsm to SDP

<--- Reliably Transmitting (no NAT) to far_end_ip:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bKYDZcf.ZZ307DDajg;received=far_end_ip
Record-Route: <sip:AAQAAZK5DAAAvAAAAvQAAfHoK@far_end_ip:5060;lr>
From: <tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To: <tel:99966;phone-context=unknown>;tag=as2dfd8460
Call-ID: 712DFF13385DB12F5FD6136D@6443ffffffff
CSeq: 1 INVITE
Server: RVT_DN_PBX_01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:99966@my_ast_ip:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 230

v=0
o=DN01 1371445892 1371445892 IN IP4 my_media_plan_ip
s=DNIVR01
c=IN IP4 my_media_plan_ip
t=0 0
m=audio 11646 RTP/AVP 8 0 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:far_end_ip:5060 --->
ACK sip:99966@my_ast_ip:5060 SIP/2.0
Content-Length:0
From:<tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To:<tel:99966;phone-context=unknown>;tag=as2dfd8460
Via:SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bK_d23V1VhgXa76_Y1
Call-ID:712DFF13385DB12F5FD6136D@6443ffffffff
CSeq:1 ACK
Max-Forwards:70
Request-Disposition:no-fork

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:far_end_ip:5060 --->
INVITE sip:99966@my_ast_ip:5060 SIP/2.0
Content-Length:149
From:<tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To:<tel:99966;phone-context=unknown>;tag=as2dfd8460
Via:SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bKYXiWaj0BXZ.U4ece
Call-ID:712DFF13385DB12F5FD6136D@6443ffffffff
CSeq:2 INVITE
Max-Forwards:70
Record-Route:<sip:AAQAAZK5DAAAvAAAAvQAAfHoK@far_end_ip:5060;lr>
Request-Disposition:no-fork
Session-Expires:1800;refresher=uac
Contact:sip:far_end_ip:5060
Supported:timer
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE
P-Charging-Vector:icid-value=aZWWkCSMvZAYAAAAEKDMBWRD3BA-;icid-generated-at=far_end_ip;orig-ioi=kw.zain.com
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 1 IN IP4 far_end_owner_ip
s=-
c=IN IP4 far_end_media_plan_ip
t=0 0
m=audio 8682 RTP/AVP 8
b=AS:80
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:20
<------------->
--- (17 headers 10 lines) ---
Sending to far_end_ip:5060 (no NAT)
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - (alaw|ulaw|gsm), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port far_end_media_plan_ip:8682

<--- Transmitting (no NAT) to far_end_ip:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bKYXiWaj0BXZ.U4ece;received=far_end_ip
Record-Route: <sip:AAQAAZK5DAAAvAAAAvQAAfHoK@far_end_ip:5060;lr>
From: <tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To: <tel:99966;phone-context=unknown>;tag=as2dfd8460
Call-ID: 712DFF13385DB12F5FD6136D@6443ffffffff
CSeq: 2 INVITE
Server: RVT_DN_PBX_01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:99966@my_ast_ip:5060>
Content-Length: 0


<------------>
Audio is at 11646
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec gsm to SDP

<--- Reliably Transmitting (no NAT) to far_end_ip:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bKYXiWaj0BXZ.U4ece;received=far_end_ip
Record-Route: <sip:AAQAAZK5DAAAvAAAAvQAAfHoK@far_end_ip:5060;lr>
From: <tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To: <tel:99966;phone-context=unknown>;tag=as2dfd8460
Call-ID: 712DFF13385DB12F5FD6136D@6443ffffffff
CSeq: 2 INVITE
Server: RVT_DN_PBX_01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:99966@my_ast_ip:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 230

v=0
o=DN01 1371445892 1371445893 IN IP4 my_media_plan_ip
s=DNIVR01
c=IN IP4 my_media_plan_ip
t=0 0
m=audio 11646 RTP/AVP 8 0 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:far_end_ip:5060 --->
ACK sip:99966@my_ast_ip:5060 SIP/2.0
Content-Length:0
From:<tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To:<tel:99966;phone-context=unknown>;tag=as2dfd8460
Via:SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bKDjYjgYb294DY8YB.
Call-ID:712DFF13385DB12F5FD6136D@6443ffffffff
CSeq:2 ACK
Max-Forwards:70
Request-Disposition:no-fork

<------------->
--- (9 headers 0 lines) ---
       > 0x7f31c4007bb0 -- Probation passed - setting RTP source address to far_end_media_plan_ip:8682
Got  RTP packet from    far_end_media_plan_ip:8682 (type 08, seq 000002, ts 041760, len 000160)
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023664, ts 000160, len 000160)
    -- <SIP/+96597834852-00000000> Playing 'hello-world.gsm' (language 'en')
Got  RTP packet from    far_end_media_plan_ip:8682 (type 08, seq 000003, ts 041920, len 000160)
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023665, ts 000320, len 000160)
Got  RTP packet from    far_end_media_plan_ip:8682 (type 08, seq 000004, ts 042080, len 000160)
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023666, ts 000480, len 000160)
.
.
.
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023738, ts 012000, len 000160)
Got  RTP packet from    far_end_media_plan_ip:8682 (type 08, seq 000077, ts 053760, len 000160)
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023739, ts 012160, len 000160)
Got  RTP packet from    far_end_media_plan_ip:8682 (type 08, seq 000078, ts 053920, len 000160)
    -- Executing [s@noanswer_demo:8] Playback("SIP/+96597834852-00000000", "demo-instruct") in new stack
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023740, ts 012320, len 000160)
    -- <SIP/+96597834852-00000000> Playing 'demo-instruct.gsm' (language 'en')
Got  RTP packet from    far_end_media_plan_ip:8682 (type 08, seq 000079, ts 054080, len 000160)
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023741, ts 012480, len 000160)
Got  RTP packet from    far_end_media_plan_ip:8682 (type 08, seq 000080, ts 054240, len 000160)
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023742, ts 012640, len 000160)
.
.
.
Got  RTP packet from    far_end_media_plan_ip:8682 (type 08, seq 000125, ts 061440, len 000160)
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023787, ts 019840, len 000160)
Got  RTP packet from    far_end_media_plan_ip:8682 (type 08, seq 000126, ts 061600, len 000160)
Sent RTP packet to      far_end_media_plan_ip:8682 (type 08, seq 023788, ts 020000, len 000160)

<--- SIP read from UDP:far_end_ip:5060 --->
BYE sip:99966@my_ast_ip:5060 SIP/2.0
Content-Length:6
From:<tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To:<tel:99966;phone-context=unknown>;tag=as2dfd8460
Via:SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bK0B5B02_.W5fZYi.6
Call-ID:712DFF13385DB12F5FD6136D@6443ffffffff
CSeq:3 BYE
Max-Forwards:70
Request-Disposition:no-fork
Supported:timer
Reason:X.int ;reasoncode=0x00000000;add-info=0132.0001.0B2E
Reason:Q.850 ;cause=16
Content-Type:application/ISUP;base=itu-t92+;version=itu-t92+
Content-Disposition:signal;handling=required
Content-Transfer-Encoding:binary



<------------->
--- (15 headers 1 lines) ---
Sending to far_end_ip:5060 (no NAT)
Scheduling destruction of SIP dialog '712DFF13385DB12F5FD6136D@6443ffffffff' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to far_end_ip:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP far_end_domain:5060;branch=z9hG4bK0B5B02_.W5fZYi.6;received=far_end_ip
From: <tel:+96597834852>;tag=CWCA8fBX1Y6h8902
To: <tel:99966;phone-context=unknown>;tag=as2dfd8460
Call-ID: 712DFF13385DB12F5FD6136D@6443ffffffff
CSeq: 3 BYE
Server: RVT_DN_PBX_01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Executing [h@noanswer_demo:1] Goto("SIP/+96597834852-00000000", "obd_example_hangup,1") in new stack
    -- Goto (noanswer_demo,obd_example_hangup,1)
Really destroying SIP dialog '712DFF13385DB12F5FD6136D@6443ffffffff' Method: BYE

any clue what I am doing wrong here ?

Modification 2015-08-25:
[color=#FF0000]I am receiving SIP-I messages and ignoring the ISUP part[/color]

I have finally solved this issue :smiley:

thanks to the post No RingBack Tone by MasterLog

progressinband=yes
internal_timing=yes
silencesuppression=no

and my Dial command :: Dial(“SIP/@,45”)

this passes any indication (received from B-Party) to A-Party.

Thank you david55 for the help.

by the way, now I am working with Asterisk 11.3.0, modified to support SIP-I.
will try to patch 13.4.0 to support SIP-I (if I get time).

How can I mark this as solved ??