Not able to connect call

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 ?

Probably you are using some codec that’s not available in your server. Set Linpgone to use G711/ulaw and repeat your process.

Hi ,
Thanks for your reply . Issue has been fixed . I have just enable AVPF . I do not know what is the AVPF what is the effect of this after enabling in sip account .

If you can give any information regarding AVPF. That will be good for me .

Thanks

What it is ?

AVPF adds some very interesting RTCP message called PLI (Packet Loss Indication), SLI (Slice Loss Indication), RPSI (Reference Picture Indication), that allow fast error recovery of video when packet transmission errors occur.

AVPF is enabled by default in Liblinphone, visible in the "a=rtcp-fb" attributes in the SDP messages. To be effectively used, both caller and callee must support it, and if the server is a back to back user agent, it shall also support it.

https://wiki.linphone.org/xwiki/wiki/public/view/Linphone/Good%20practices%20for%20using%20SIP/

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