Hi ,
I am trying to call with Linphone android but getting incompatible midea parameter error on mobile side .
Asterisk Log
<— Transmitting SIP request (427 bytes) to UDP:157.39.2.66:59357 —>
OPTIONS sip:101224@157.39.2.66:59357 SIP/2.0
Via: SIP/2.0/UDP 15.206.90.85:8080;rport;branch=z9hG4bKPj478592ae-9142-412f-83f8-f3feeedfd384
From: sip:101224@172.31.4.130;tag=de796001-a832-4cee-abb4-7577164c43ca
To: sip:101224@157.39.2.66
Contact: sip:101224@15.206.90.85:8080
Call-ID: ba1572ba-f4b1-48b6-b9de-a8e23def0f84
CSeq: 28431 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0
<— Transmitting SIP request (427 bytes) to UDP:157.39.2.66:59357 —>
OPTIONS sip:101224@157.39.2.66:59357 SIP/2.0
Via: SIP/2.0/UDP 15.206.90.85:8080;rport;branch=z9hG4bKPj478592ae-9142-412f-83f8-f3feeedfd384
From: sip:101224@172.31.4.130;tag=de796001-a832-4cee-abb4-7577164c43ca
To: sip:101224@157.39.2.66
Contact: sip:101224@15.206.90.85:8080
Call-ID: ba1572ba-f4b1-48b6-b9de-a8e23def0f84
CSeq: 28431 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0
<— Received SIP request (1107 bytes) from UDP:157.39.2.66:59357 —>
INVITE sip:101223@15.206.90.85 SIP/2.0
Via: SIP/2.0/UDP 25.80.210.129:59357;branch=z9hG4bK.SVUIl3xN5;rport
From: sip:101224@15.206.90.85;tag=qElu6S08J
To: sip:101223@15.206.90.85
CSeq: 20 INVITE
Call-ID: xqc-RWWhOL
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 475
Contact: sip:101224@157.39.2.66:59357;transport=udp;+sip.instance=“urn:uuid:e243dca2-8da8-000f-bbd8-e328bfeebf6f”
User-Agent: LinphoneAndroid/4.2.1 (Redmi) LinphoneSDK/4.3.1-pre.12+516dacf (release/4.3)
v=0
o=101224 3425 337 IN IP4 25.80.210.129
s=Talk
c=IN IP4 25.80.210.129
t=0 0
a=ice-pwd:b42029ad92bfdf9fa6c88797
a=ice-ufrag:ecc82c50
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 3 9 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=candidate:1 1 UDP 2130706303 25.80.210.129 7078 typ host
a=candidate:1 2 UDP 2130706302 25.80.210.129 7079 typ host
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
<— Transmitting SIP response (458 bytes) to UDP:157.39.2.66:59357 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 25.80.210.129:59357;rport=59357;received=157.39.2.66;branch=z9hG4bK.SVUIl3xN5
Call-ID: xqc-RWWhOL
From: sip:101224@15.206.90.85;tag=qElu6S08J
To: sip:101223@15.206.90.85;tag=z9hG4bK.SVUIl3xN5
CSeq: 20 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1582951578/006db15ae60280787c899ba06c123cdb”,opaque=“35e1607d5adb1e0d”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 15.6.1
Content-Length: 0
<— Received SIP request (380 bytes) from UDP:157.39.2.66:59357 —>
ACK sip:101223@15.206.90.85 SIP/2.0
Via: SIP/2.0/UDP 25.80.210.129:59357;branch=z9hG4bK.SVUIl3xN5;rport
Call-ID: xqc-RWWhOL
From: sip:101224@15.206.90.85;tag=qElu6S08J
To: sip:101223@15.206.90.85;tag=z9hG4bK.SVUIl3xN5
Contact: sip:101224@157.39.2.66:59357;transport=udp;+sip.instance=“urn:uuid:e243dca2-8da8-000f-bbd8-e328bfeebf6f”
Max-Forwards: 70
CSeq: 20 ACK
<— Received SIP request (1389 bytes) from UDP:157.39.2.66:59357 —>
INVITE sip:101223@15.206.90.85 SIP/2.0
Via: SIP/2.0/UDP 25.80.210.129:59357;branch=z9hG4bK.if1hYdoHW;rport
From: sip:101224@15.206.90.85;tag=qElu6S08J
To: sip:101223@15.206.90.85
CSeq: 21 INVITE
Call-ID: xqc-RWWhOL
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 475
Contact: sip:101224@157.39.2.66:59357;transport=udp;+sip.instance=“urn:uuid:e243dca2-8da8-000f-bbd8-e328bfeebf6f”
User-Agent: LinphoneAndroid/4.2.1 (Redmi) LinphoneSDK/4.3.1-pre.12+516dacf (release/4.3)
Authorization: Digest realm=“asterisk”, nonce=“1582951578/006db15ae60280787c899ba06c123cdb”, algorithm=md5, opaque=“35e1607d5adb1e0d”, username=“101224”, uri="sip:101223@15.206.90.85", response=“e7a06dea13239bcfca57af851cdd0273”, cnonce=“0zK8JNXKQ4vL0axp”, nc=00000001, qop=auth
v=0
o=101224 3425 337 IN IP4 25.80.210.129
s=Talk
c=IN IP4 25.80.210.129
t=0 0
a=ice-pwd:b42029ad92bfdf9fa6c88797
a=ice-ufrag:ecc82c50
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 3 9 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=candidate:1 1 UDP 2130706303 25.80.210.129 7078 typ host
a=candidate:1 2 UDP 2130706302 25.80.210.129 7079 typ host
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
== Setting global variable ‘SIPDOMAIN’ to ‘15.206.90.85’
<— Transmitting SIP response (283 bytes) to UDP:157.39.2.66:59357 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 25.80.210.129:59357;rport=59357;received=157.39.2.66;branch=z9hG4bK.if1hYdoHW
Call-ID: xqc-RWWhOL
From: sip:101224@15.206.90.85;tag=qElu6S08J
To: sip:101223@15.206.90.85
CSeq: 21 INVITE
Server: Asterisk PBX 15.6.1
Content-Length: 0
<— Transmitting SIP response (337 bytes) to UDP:157.39.2.66:59357 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 25.80.210.129:59357;rport=59357;received=157.39.2.66;branch=z9hG4bK.if1hYdoHW
Call-ID: xqc-RWWhOL
From: sip:101224@15.206.90.85;tag=qElu6S08J
To: sip:101223@15.206.90.85;tag=5a4a8f40-98bc-4b62-88b3-1799698d23a0
CSeq: 21 INVITE
Server: Asterisk PBX 15.6.1
Content-Length: 0
<— Received SIP request (399 bytes) from UDP:157.39.2.66:59357 —>
ACK sip:101223@15.206.90.85 SIP/2.0
Via: SIP/2.0/UDP 25.80.210.129:59357;branch=z9hG4bK.if1hYdoHW;rport
Call-ID: xqc-RWWhOL
From: sip:101224@15.206.90.85;tag=qElu6S08J
To: sip:101223@15.206.90.85;tag=5a4a8f40-98bc-4b62-88b3-1799698d23a0
Contact: sip:101224@157.39.2.66:59357;transport=udp;+sip.instance=“urn:uuid:e243dca2-8da8-000f-bbd8-e328bfeebf6f”
Max-Forwards: 70
CSeq: 21 ACK
<— Transmitting SIP request (427 bytes) to UDP:157.39.2.66:59357 —>
OPTIONS sip:101224@157.39.2.66:59357 SIP/2.0
Via: SIP/2.0/UDP 15.206.90.85:8080;rport;branch=z9hG4bKPj478592ae-9142-412f-83f8-f3feeedfd384
From: sip:101224@172.31.4.130;tag=de796001-a832-4cee-abb4-7577164c43ca
To: sip:101224@157.39.2.66
Contact: sip:101224@15.206.90.85:8080
Call-ID: ba1572ba-f4b1-48b6-b9de-a8e23def0f84
CSeq: 28431 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.6.1
Content-Length: 0
<— Received SIP response (295 bytes) from UDP:157.39.2.66:59357 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 15.206.90.85:8080;rport;branch=z9hG4bKPj478592ae-9142-412f-83f8-f3feeedfd384
From: sip:101224@172.31.4.130;tag=de796001-a832-4cee-abb4-7577164c43ca
To: sip:101224@157.39.2.66;tag=cB64Q
Call-ID: ba1572ba-f4b1-48b6-b9de-a8e23def0f84
CSeq: 28431 OPTIONS
ip-172-31-4-130*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Please help . What is the wrong with this ?