Intercom (Akuvox R20K) Early Media

Asterisk Asterisk SIP
1

Hello all,

i’m trying to get my Intercom to send Early Media while calling linphone on my smartphone (android).

Here is the relevant part of my DialPlan:
exten => 666,1,Progress()
exten => 666,n,Wait(2)
exten => 666,n,Dial(PJSIP/6001,a)
exten => 666,n,Answer()
exten => 666,n,Hangup

The call works and i get video after answering. But not before. (Early media is enabled in Linphone)

Here is a Log of PJSIP while making the Call. Cann anyone guide me to make this work?

    -- Added contact 'sip:6001@192.168.2.114:41290;transport=udp' to AOR '6001' with expiration of 3600 seconds
  == Endpoint 6001 is now Reachable
    -- Executing [666@internal:1] Progress("PJSIP/6002-00000000", "") in new stack
    -- Executing [666@internal:2] Wait("PJSIP/6002-00000000", "2") in new stack
    -- Executing [666@internal:3] Dial("PJSIP/6002-00000000", "PJSIP/6001,,a") in new stack
    -- Called PJSIP/6001
    -- PJSIP/6001-00000001 is making progress passing it to PJSIP/6002-00000000
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [666@internal:4] Answer("PJSIP/6002-00000000", "") in new stack
    -- Executing [666@internal:5] Hangup("PJSIP/6002-00000000", "") in new stack
  == Spawn extension (internal, 666, 5) exited non-zero on 'PJSIP/6002-00000000'
archlinux*CLI> pjsip set logger on
PJSIP Logging enabled
archlinux*CLI> sip set debug on
No such command 'sip set debug on' (type 'core show help sip set' for other possible commands)
<--- Received SIP request (1309 bytes) from UDP:192.168.2.132:5062 --->
INVITE sip:666@192.168.2.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.132:5062;rport;branch=z9hG4bK1257799794
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>
Call-ID: 569681582@192.168.2.132
CSeq: 20 INVITE
Contact: <sip:6002@192.168.2.132:5062>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: ABC
Subject: call invite
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length:   721

v=0
o=6002 5000 5000 IN IP4 192.168.2.132
s=Talk
c=IN IP4 192.168.2.132
b=AS:4000
t=0 0
m=audio 11972 RTP/AVP 0 8 9 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:OTA1MTAxNDUzMDIwOTAwNjM3NTYwMDkwNTE2MDYw
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OTA1MTAxNDUzMDIwOTAwNjM3NTYwMDkwNTE2MDYw
a=crypto:3 F8_128_HMAC_SHA1_80 inline:OTA1MTAxNDUzMDIwOTAwNjM3NTYwMDkwNTE2MDYw
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 11974 RTP/AVP 99
a=ptime:20
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42e01e;packetization-mode=1;max-br=2048;max-mbps=40500
a=sendrecv

<--- Transmitting SIP response (479 bytes) to UDP:192.168.2.132:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.132:5062;rport=5062;received=192.168.2.132;branch=z9hG4bK1257799794
Call-ID: 569681582@192.168.2.132
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>;tag=z9hG4bK1257799794
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1711532889/54f6da31fe02832e0295f1bdf94bd3d7",opaque="43f90fb478865184",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.2
Content-Length:  0


<--- Received SIP request (298 bytes) from UDP:192.168.2.132:5062 --->
ACK sip:666@192.168.2.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.132:5062;rport;branch=z9hG4bK1257799794
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>;tag=z9hG4bK1257799794
Call-ID: 569681582@192.168.2.132
CSeq: 20 ACK
User-Agent: ABC
Content-Length: 0


<--- Received SIP request (1577 bytes) from UDP:192.168.2.132:5062 --->
INVITE sip:666@192.168.2.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.132:5062;rport;branch=z9hG4bK1990689052
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>
Call-ID: 569681582@192.168.2.132
CSeq: 21 INVITE
Contact: <sip:6002@192.168.2.132:5062>
Authorization: Digest username="6002", realm="asterisk", nonce="1711532889/54f6da31fe02832e0295f1bdf94bd3d7", uri="sip:666@192.168.2.133", response="aafdcea24b61099ce583d8b426d01fd7", algorithm=MD5, cnonce="0a4f113b", opaque="43f90fb478865184", qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: ABC
Subject: call invite
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length:   721

v=0
o=6002 5000 5000 IN IP4 192.168.2.132
s=Talk
c=IN IP4 192.168.2.132
b=AS:4000
t=0 0
m=audio 11972 RTP/AVP 0 8 9 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:OTA1MTAxNDUzMDIwOTAwNjM3NTYwMDkwNTE2MDYw
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:OTA1MTAxNDUzMDIwOTAwNjM3NTYwMDkwNTE2MDYw
a=crypto:3 F8_128_HMAC_SHA1_80 inline:OTA1MTAxNDUzMDIwOTAwNjM3NTYwMDkwNTE2MDYw
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 11974 RTP/AVP 99
a=ptime:20
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42e01e;packetization-mode=1;max-br=2048;max-mbps=40500
a=sendrecv

<--- Transmitting SIP response (305 bytes) to UDP:192.168.2.132:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.132:5062;rport=5062;received=192.168.2.132;branch=z9hG4bK1990689052
Call-ID: 569681582@192.168.2.132
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>
CSeq: 21 INVITE
Server: Asterisk PBX 20.5.2
Content-Length:  0


    -- Executing [666@internal:1] Progress("PJSIP/6002-00000002", "") in new stack
    -- Executing [666@internal:2] Wait("PJSIP/6002-00000002", "2") in new stack
<--- Transmitting SIP response (927 bytes) to UDP:192.168.2.132:5062 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.132:5062;rport=5062;received=192.168.2.132;branch=z9hG4bK1990689052
Call-ID: 569681582@192.168.2.132
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>;tag=xts-eTNqU7Msea3sCSN8z7YzpMeT.OY8
CSeq: 21 INVITE
Server: Asterisk PBX 20.5.2
Contact: <sip:192.168.2.133:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   395

v=0
o=- 5000 5002 IN IP4 192.168.2.133
s=Asterisk
c=IN IP4 192.168.2.133
t=0 0
m=audio 20250 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:230
a=sendrecv
m=video 13072 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 max-mbps=40500;max-br=2048;packetization-mode=1;profile-level-id=42E01E
a=sendrecv

    -- Executing [666@internal:3] Dial("PJSIP/6002-00000002", "PJSIP/6001,,a") in new stack
    -- Called PJSIP/6001
<--- Transmitting SIP request (1109 bytes) to UDP:192.168.2.114:41290 --->
INVITE sip:6001@192.168.2.114:41290;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bKPjJfBmTD8lur1mYG9z7YJBOQu-HpEpUY8R
From: "FritzBox" <sip:6002@192.168.2.133>;tag=cXVnRuEWPU4Fl9MEoXY9DOczO5WVAN1D
To: <sip:6001@192.168.2.114>
Contact: <sip:asterisk@192.168.2.133:5060>
Call-ID: VbijubkqsmWOdHZNWWsOFE5dZ4wlKWv2
CSeq: 29515 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.2
Content-Type: application/sdp
Content-Length:   430

v=0
o=- 1531522901 1531522901 IN IP4 192.168.2.133
s=Asterisk
c=IN IP4 192.168.2.133
t=0 0
m=audio 20564 RTP/AVP 18 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:230
a=sendrecv
m=video 13428 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 max-mbps=40500;max-br=2048;packetization-mode=1;profile-level-id=42E01E
a=sendrecv

<--- Transmitting SIP request (1109 bytes) to UDP:192.168.2.114:41290 --->
INVITE sip:6001@192.168.2.114:41290;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bKPjJfBmTD8lur1mYG9z7YJBOQu-HpEpUY8R
From: "FritzBox" <sip:6002@192.168.2.133>;tag=cXVnRuEWPU4Fl9MEoXY9DOczO5WVAN1D
To: <sip:6001@192.168.2.114>
Contact: <sip:asterisk@192.168.2.133:5060>
Call-ID: VbijubkqsmWOdHZNWWsOFE5dZ4wlKWv2
CSeq: 29515 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.2
Content-Type: application/sdp
Content-Length:   430

v=0
o=- 1531522901 1531522901 IN IP4 192.168.2.133
s=Asterisk
c=IN IP4 192.168.2.133
t=0 0
m=audio 20564 RTP/AVP 18 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:230
a=sendrecv
m=video 13428 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 max-mbps=40500;max-br=2048;packetization-mode=1;profile-level-id=42E01E
a=sendrecv

<--- Received SIP response (285 bytes) from UDP:192.168.2.114:41290 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bKPjJfBmTD8lur1mYG9z7YJBOQu-HpEpUY8R
From: "FritzBox" <sip:6002@192.168.2.133>;tag=cXVnRuEWPU4Fl9MEoXY9DOczO5WVAN1D
To: sip:6001@192.168.2.114
Call-ID: VbijubkqsmWOdHZNWWsOFE5dZ4wlKWv2
CSeq: 29515 INVITE


<--- Received SIP response (801 bytes) from UDP:192.168.2.114:41290 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bKPjJfBmTD8lur1mYG9z7YJBOQu-HpEpUY8R
From: "FritzBox" <sip:6002@192.168.2.133>;tag=cXVnRuEWPU4Fl9MEoXY9DOczO5WVAN1D
To: <sip:6001@192.168.2.114>;tag=2evzLia
Call-ID: VbijubkqsmWOdHZNWWsOFE5dZ4wlKWv2
CSeq: 29515 INVITE
User-Agent: LinphoneAndroid/5.2.3 (Pixel 6) LinphoneSDK/5.3.14 (tags/5.3.14^0)
Supported: replaces, outbound, gruu, path, record-aware
Content-Type: application/sdp
Content-Length: 303

v=0
o=6001 988 772 IN IP4 192.168.2.114
s=Talk
c=IN IP4 192.168.2.114
t=0 0
m=audio 36704 RTP/AVP 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=rtcp:39097
m=video 55708 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F; packetization-mode=1
a=rtcp:51920

    -- PJSIP/6001-00000003 is making progress passing it to PJSIP/6002-00000002
<--- Transmitting SIP response (927 bytes) to UDP:192.168.2.132:5062 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.132:5062;rport=5062;received=192.168.2.132;branch=z9hG4bK1990689052
Call-ID: 569681582@192.168.2.132
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>;tag=xts-eTNqU7Msea3sCSN8z7YzpMeT.OY8
CSeq: 21 INVITE
Server: Asterisk PBX 20.5.2
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.133:5060>
Content-Type: application/sdp
Content-Length:   395

v=0
o=- 5000 5002 IN IP4 192.168.2.133
s=Asterisk
c=IN IP4 192.168.2.133
t=0 0
m=audio 20250 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:230
a=sendrecv
m=video 13072 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 max-mbps=40500;max-br=2048;packetization-mode=1;profile-level-id=42E01E
a=sendrecv

<--- Received SIP response (801 bytes) from UDP:192.168.2.114:41290 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bKPjJfBmTD8lur1mYG9z7YJBOQu-HpEpUY8R
From: "FritzBox" <sip:6002@192.168.2.133>;tag=cXVnRuEWPU4Fl9MEoXY9DOczO5WVAN1D
To: <sip:6001@192.168.2.114>;tag=2evzLia
Call-ID: VbijubkqsmWOdHZNWWsOFE5dZ4wlKWv2
CSeq: 29515 INVITE
User-Agent: LinphoneAndroid/5.2.3 (Pixel 6) LinphoneSDK/5.3.14 (tags/5.3.14^0)
Supported: replaces, outbound, gruu, path, record-aware
Content-Type: application/sdp
Content-Length: 303

v=0
o=6001 988 772 IN IP4 192.168.2.114
s=Talk
c=IN IP4 192.168.2.114
t=0 0
m=audio 36704 RTP/AVP 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=rtcp:39097
m=video 55708 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F; packetization-mode=1
a=rtcp:51920

    -- PJSIP/6001-00000003 is making progress passing it to PJSIP/6002-00000002
<--- Transmitting SIP response (927 bytes) to UDP:192.168.2.132:5062 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.132:5062;rport=5062;received=192.168.2.132;branch=z9hG4bK1990689052
Call-ID: 569681582@192.168.2.132
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>;tag=xts-eTNqU7Msea3sCSN8z7YzpMeT.OY8
CSeq: 21 INVITE
Server: Asterisk PBX 20.5.2
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.133:5060>
Content-Type: application/sdp
Content-Length:   395

v=0
o=- 5000 5002 IN IP4 192.168.2.133
s=Asterisk
c=IN IP4 192.168.2.133
t=0 0
m=audio 20250 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:230
a=sendrecv
m=video 13072 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 max-mbps=40500;max-br=2048;packetization-mode=1;profile-level-id=42E01E
a=sendrecv

<--- Received SIP request (963 bytes) from UDP:192.168.2.114:41290 --->
PUBLISH sip:6001@192.168.2.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.114:41290;branch=z9hG4bK.Tohufe~kV;rport
From: <sip:6001@192.168.2.133>;tag=sFuftD-JZ
To: sip:6001@192.168.2.133
CSeq: 20 PUBLISH
Call-ID: rMNopBGYM0
Max-Forwards: 70
Supported: replaces, outbound, gruu, path, record-aware
Event: presence
Content-Type: application/pidf+xml
Content-Length: 495
Expires: 3600
User-Agent: LinphoneAndroid/5.2.3 (Pixel 6) LinphoneSDK/5.3.14 (tags/5.3.14^0)

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:pidfonline="http://www.linphone.org/xsds/pidfonline.xsd" entity="sip:6001@192.168.2.133" xmlns="urn:ietf:params:xml:ns:pidf">
 <tuple id="_gvm1k">
  <status>
   <basic>open</basic>
   <pidfonline:online/>
  </status>
  <contact priority="0.8">sip:6001@192.168.2.133</contact>
  <timestamp>2024-03-27T09:48:21Z</timestamp>
 </tuple>
</presence>

<--- Transmitting SIP response (458 bytes) to UDP:192.168.2.114:41290 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.114:41290;rport=41290;received=192.168.2.114;branch=z9hG4bK.Tohufe~kV
Call-ID: rMNopBGYM0
From: <sip:6001@192.168.2.133>;tag=sFuftD-JZ
To: <sip:6001@192.168.2.133>;tag=z9hG4bK.Tohufe~kV
CSeq: 20 PUBLISH
WWW-Authenticate: Digest realm="asterisk",nonce="1711532900/74036964a1d26547a9c9a45d4d5887b7",opaque="2d439ab93b26e45a",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.2
Content-Length:  0


<--- Received SIP request (1242 bytes) from UDP:192.168.2.114:41290 --->
PUBLISH sip:6001@192.168.2.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.114:41290;branch=z9hG4bK.p4h0k3qxO;rport
From: <sip:6001@192.168.2.133>;tag=sFuftD-JZ
To: sip:6001@192.168.2.133
CSeq: 21 PUBLISH
Call-ID: rMNopBGYM0
Max-Forwards: 70
Supported: replaces, outbound, gruu, path, record-aware
Event: presence
Content-Type: application/pidf+xml
Content-Length: 495
Expires: 3600
User-Agent: LinphoneAndroid/5.2.3 (Pixel 6) LinphoneSDK/5.3.14 (tags/5.3.14^0)
Authorization:  Digest realm="asterisk", nonce="1711532900/74036964a1d26547a9c9a45d4d5887b7", algorithm=MD5, opaque="2d439ab93b26e45a", username="6001",  uri="sip:6001@192.168.2.133", response="1120e2126942ad0eb2bc7aff9345af48", cnonce="~PyszhSqkQFqHj2J", nc=00000001, qop=auth

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:pidfonline="http://www.linphone.org/xsds/pidfonline.xsd" entity="sip:6001@192.168.2.133" xmlns="urn:ietf:params:xml:ns:pidf">
 <tuple id="_gvm1k">
  <status>
   <basic>open</basic>
   <pidfonline:online/>
  </status>
  <contact priority="0.8">sip:6001@192.168.2.133</contact>
  <timestamp>2024-03-27T09:48:21Z</timestamp>
 </tuple>
</presence>

[Mar 27 10:48:20] WARNING[290]: res_pjsip_pubsub.c:3396 pubsub_on_rx_publish_request: No registered publish handler for event presence from 6001
<--- Transmitting SIP response (309 bytes) to UDP:192.168.2.114:41290 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.2.114:41290;rport=41290;received=192.168.2.114;branch=z9hG4bK.p4h0k3qxO
Call-ID: rMNopBGYM0
From: <sip:6001@192.168.2.133>;tag=sFuftD-JZ
To: <sip:6001@192.168.2.133>;tag=z9hG4bK.p4h0k3qxO
CSeq: 21 PUBLISH
Server: Asterisk PBX 20.5.2
Content-Length:  0


<--- Received SIP response (437 bytes) from UDP:192.168.2.114:41290 --->
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bKPjJfBmTD8lur1mYG9z7YJBOQu-HpEpUY8R
From: "FritzBox" <sip:6002@192.168.2.133>;tag=cXVnRuEWPU4Fl9MEoXY9DOczO5WVAN1D
To: <sip:6001@192.168.2.114>;tag=2evzLia
Call-ID: VbijubkqsmWOdHZNWWsOFE5dZ4wlKWv2
CSeq: 29515 INVITE
User-Agent: LinphoneAndroid/5.2.3 (Pixel 6) LinphoneSDK/5.3.14 (tags/5.3.14^0)
Supported: replaces, outbound, gruu, path, record-aware


<--- Transmitting SIP request (403 bytes) to UDP:192.168.2.114:41290 --->
ACK sip:6001@192.168.2.114:41290;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bKPjJfBmTD8lur1mYG9z7YJBOQu-HpEpUY8R
From: "FritzBox" <sip:6002@192.168.2.133>;tag=cXVnRuEWPU4Fl9MEoXY9DOczO5WVAN1D
To: <sip:6001@192.168.2.114>;tag=2evzLia
Call-ID: VbijubkqsmWOdHZNWWsOFE5dZ4wlKWv2
CSeq: 29515 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.2
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [666@internal:4] Answer("PJSIP/6002-00000002", "") in new stack
<--- Transmitting SIP response (961 bytes) to UDP:192.168.2.132:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.132:5062;rport=5062;received=192.168.2.132;branch=z9hG4bK1990689052
Call-ID: 569681582@192.168.2.132
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>;tag=xts-eTNqU7Msea3sCSN8z7YzpMeT.OY8
CSeq: 21 INVITE
Server: Asterisk PBX 20.5.2
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.2.133:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   395

v=0
o=- 5000 5002 IN IP4 192.168.2.133
s=Asterisk
c=IN IP4 192.168.2.133
t=0 0
m=audio 20250 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:230
a=sendrecv
m=video 13072 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 max-mbps=40500;max-br=2048;packetization-mode=1;profile-level-id=42E01E
a=sendrecv

<--- Received SIP request (372 bytes) from UDP:192.168.2.132:5062 --->
ACK sip:192.168.2.133:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.132:5062;rport;branch=z9hG4bK1131117363
From: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
To: <sip:666@192.168.2.133>;tag=xts-eTNqU7Msea3sCSN8z7YzpMeT.OY8
Call-ID: 569681582@192.168.2.132
CSeq: 21 ACK
Contact: <sip:6002@192.168.2.132:5062>
Max-Forwards: 70
User-Agent: ABC
Content-Length: 0


    -- Executing [666@internal:5] Hangup("PJSIP/6002-00000002", "") in new stack
  == Spawn extension (internal, 666, 5) exited non-zero on 'PJSIP/6002-00000002'
<--- Transmitting SIP request (405 bytes) to UDP:192.168.2.132:5062 --->
BYE sip:6002@192.168.2.132:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bKPjU6xrhVN.Y0V.bJ7xmNIn2h1x4IhGeDlh
From: <sip:666@192.168.2.133>;tag=xts-eTNqU7Msea3sCSN8z7YzpMeT.OY8
To: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
Call-ID: 569681582@192.168.2.132
CSeq: 28796 BYE
Reason: Q.850;cause=21
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.2
Content-Length:  0


<--- Received SIP response (326 bytes) from UDP:192.168.2.132:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.133:5060;rport=5060;branch=z9hG4bKPjU6xrhVN.Y0V.bJ7xmNIn2h1x4IhGeDlh
From: <sip:666@192.168.2.133>;tag=xts-eTNqU7Msea3sCSN8z7YzpMeT.OY8
To: "FritzBox" <sip:6002@192.168.2.133>;tag=1751597088
Call-ID: 569681582@192.168.2.132
CSeq: 28796 BYE
User-Agent: ABC
Content-Length: 0
2

Some Progress:

i can make it work with this:
exten => 666,1,Progress()
exten => 666,n,Answer(500)
exten => 666,n,Dial(PJSIP/6001)
exten => 666,n,Hangup

Linphone now shows the video even before i answer the call. Great.

The Problem no is that there is no “ringing” tone on the intercom anymore.

With
exten => 666,1,Progress()
exten => 666,n,Answer(500)
exten => 666,n,Ringing()
exten => 666,n,Wait(5)
exten => 666,n,Dial(PJSIP/6001)
exten => 666,n,Hangup

it is ringing for 5 seconds. But the Dial command interrupts the ringing.

How can i keep the intercom ringing until i answer the call in linphone?

3

There is a good chance you will break early media, but you could try the r option, and, if that doesn’t work, use the & notation to call a local channel, that doesn’t answer, in parallel.

I was actually surprised that you got any early media at all, as other questions I’ve seen on video doorphones, suggested that video was never supported for early media. Maybe that was only in the other direction.

By ringing, I’m assuming you mean ringback tone.

4

Yes i mean the ringing Sound in the intercom (call goes from intercom to my Phone)

I do Not really understand the r Option (you mean om the dial command right)?

Can you Provider a Sample?

5

you are right…

with

exten => 666,1,Progress()
exten => 666,n,Answer(500)
exten => 666,n,Dial(PJSIP/6001,r)
exten => 666,n,Hangup

the ringing on the intercom is back… but early video does not work…

with

exten => 666,1,Progress()
exten => 666,n,Answer(500)
exten => 666,n,Dial(PJSIP/6001,)
exten => 666,n,Hangup

video works but no ringing on the intercom

can you explain what you meant with the & notation

6

another try.

even using MOH breaks the early video stream:

exten => 666,1,Progress()
exten => 666,n,Answer(500)
exten => 666,n,Dial(PJSIP/6001,m(mohmp3))
exten => 666,n,Hangup

the music file is played on the intercom. but early video is then broken…

any other ideas?

7

On Wednesday 27 March 2024 at 17:30:55, fracoon via Asterisk Community wrote:

any other ideas?

My understanding of early media is that if you enable it, then both the audio
and the video must come from the same source - Asterisk can’t supply the audio
(ringing) whilst the doorphone supplies the video (image).

So, either your audio comes from Asterisk (you hear ringing) but there is no
image (because Asterisk doesn’t generate video), or else your video comes from
the doorphone (early mdeia) and you hear no ringing (because the doorphone
doesn’t send any ringing sound).

I don’t believe it’s possible to mix audio from one source (Asterisk) plus
video from another (the doorphone), whether you’re using early media or not.

Antony.

–
Behind the counter a boy with a shaven head stared vacantly into space,
a dozen spikes of microsoft protruding from the socket behind his ear.

  • William Gibson, Neuromancer (1984)

                                                Please reply to the list;
                                                  please *don't* CC me.
8

I think you’ve missed a detail. I think the OP wants early media video to go to the callee and ringback to the caller. They want their visitor be comforted by the ringback tone, whilst themselves being able to preview the visitor and not even admit to being in, if they don’t like them.

9

I actually meant R, on Dial, but r is possibly better, for your application. However, I think it is handled much like m, for early media.

10

On Wednesday 27 March 2024 at 18:07:06, david551 via Asterisk Community wrote:

I think you’ve missed a detail. I think the OP wants early media video to
go to the callee and ringback to the caller.

Aha! You’re right :slight_smile: Oops…

Antony.

–
Late in 1972 President Richard Nixon announced that the rate of increase of
inflation was decreasing. This was the first time a sitting president used a
third derivative to advance his case for re-election.

  • Hugo Rossi, Notices of the American Mathematical Society

                                                Please reply to the list;
                                                  please *don't* CC me.
11

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