Error 488 possible codec issue?

Asterisk Asterisk SIP
1

l have allow all the codec when l was creating endpoint, but l see this in the CLI output

Capture
Capture1140×223 196 KB

when l tried to make call, the call just does not go through(l was using TLS, but l changed back to UDP, same error 488), and l see the following in my android studio. My research indicates it is due to unsupported codec, but the log showed that things like g722 is added to possible list of codec. What is going on with this error 488?

I/: libmswebrtc 1.1.1 plugin loaded, iSAC codec version 3.6.0, iLBC codec version 1.1.1
Plugin loaded (libmswebrtc.so)
All plugins in list correctly loaded
Core callbacks [0x6fd32c8b90] registered on core [0x6f7cc45c00]
oRTP-4.3.0 initialized.
Codec opus/48000 fmtp=[useinbandfec=1] number=-1, default enablement: 16) added to the list of possible codecs.
Codec SILK/16000 fmtp= number=-1, default enablement: 16) added to the list of possible codecs.
Codec speex/16000 fmtp=[vbr=on] number=-1, default enablement: 16) added to the list of possible codecs.
Codec speex/8000 fmtp=[vbr=on] number=-1, default enablement: 16) added to the list of possible codecs.
Codec PCMU/8000 fmtp= number=0, default enablement: 16) added to the list of possible codecs.
Codec PCMA/8000 fmtp= number=8, default enablement: 16) added to the list of possible codecs.
Codec red/1000 fmtp= number=-1, default enablement: 16) added to the list of possible codecs.
Codec t140/1000 fmtp= number=-1, default enablement: 16) added to the list of possible codecs.
Codec GSM/8000 fmtp= number=3, default enablement: 0) added to the list of possible codecs.
Codec G722/8000 fmtp= number=9, default enablement: 0) added to the list of possible codecs.
Codec iLBC/8000 fmtp=[mode=30] number=-1, default enablement: 0) added to the list of possible codecs.
Codec AMR/8000 fmtp=[octet-align=1] number=-1, default enablement: 0) added to the list of possible codecs.
Codec AMR-WB/16000 fmtp=[octet-align=1] number=-1, default enablement: 0) added to the list of possible codecs.
Codec G729/8000 fmtp=[annexb=yes] number=18, default enablement: 16) added to the list of possible codecs.
Codec mpeg4-generic/16000 fmtp=[config=F8EE2000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5] number=-1, default enablement: 0) added to the list of possible codecs.
Codec mpeg4-generic/22050 fmtp=[config=F8EE2000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5] number=-1, default enablement: 0) added to the list of possible codecs.
Codec mpeg4-generic/32000 fmtp=[config=F8E82000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5] number=-1, default enablement: 0) added to the list of possible codecs.
Codec mpeg4-generic/44100 fmtp=[config=F8E82000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5] number=-1, default enablement: 0) added to the list of possible codecs.
Codec mpeg4-generic/48000 fmtp=[config=F8EE2000; constantDuration=512; indexDeltaLength=3; indexLength=3; mode=AAC-hbr; profile-level-id=76; sizeLength=13; streamType=5] number=-1, default enablement: 0) added to the list of possible codecs.
Codec iSAC/16000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec speex/32000 fmtp=[vbr=on] number=-1, default enablement: 0) added to the list of possible codecs.
Codec SILK/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec SILK/12000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec SILK/24000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec G726-16/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec G726-24/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec G726-32/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec G726-40/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec AAL2-G726-16/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec AAL2-G726-24/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec AAL2-G726-32/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec AAL2-G726-40/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec CODEC2/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec BV16/8000 fmtp= number=-1, default enablement: 0) added to the list of possible codecs.
Codec 1016/8000 fmtp= number=1, default enablement: 0) added to the list of possible codecs.
Codec G723/8000 fmtp= number=4, default enablement: 0) added to the list of possible codecs.
Codec LPC/8000 fmtp= number=7, default enablement: 0) added to the list of possible codecs.
Codec L16/44100 fmtp= number=10, default enablement: 0) added to the list of possible codecs.
Codec L16/44100 fmtp= number=11, default enablement: 0) added to the list of possible codecs.
Codec CN/8000 fmtp= number=13, default enablement: 0) added to the list of possible codecs.

I/ViewRootImpl@91ce36b[DialingActivity]: ViewPostIme pointer 0
I/ViewRootImpl@91ce36b[DialingActivity]: ViewPostIme pointer 1
I/Dialer: New MediaSession [0x6fd335f1d8] initialized (LinphoneCore version: 4.3)
I/Dialer: Found media local-ip from signaling.
CallSession [0x6fd335f1d8], stream type [audio], multicast role is [inactive]
RtpSession bound to [::0] ports [7078] [7079]
rtp_session_enable_network_simulation:DISABLING NETWORK SIMULATION
Configured srtp crypto suite: AES_CM_128_HMAC_SHA1_80
Configured srtp crypto suite: AES_CM_128_HMAC_SHA1_32
Configured srtp crypto suite: AES_256_CM_HMAC_SHA1_80
Configured srtp crypto suite: AES_256_CM_HMAC_SHA1_32
Creating ZRTP engine on rtp session [0x6f7ceea000] ssrc 0xe5164d53
Setting DSCP to 46 for MSAudio stream.
Equalizer location: hp
cannot set noise gate mode to [0] because no volume send
CallSession [0x6fd335f1d8], stream type [text], multicast role is [inactive]
I/Dialer: RtpSession bound to [::0] ports [11078] [11079]
rtp_session_enable_network_simulation:DISABLING NETWORK SIMULATION
Linphone core [0x6f7cc45c00] notified [call_created]
CallSession [0x6fd335f1d8] moving from state State::Idle to State::OutgoingInit

I/Dialer: class mil.navy.nosis.dialer.services.LinphoneServiceTEST LOG I
Linphone core [0x6f7cc45c00] notified [call_state_changed]
Found media local-ip from signaling.
Don’t put video stream on local offer for CallSession [0x6fd335f1d8]
Don’t put text stream on local offer for CallSession [0x6fd335f1d8]
I/Dialer: ms_filter_link: MSRtpRecv:0x6f7cf95900,0–>MSVoidSink:0x6f7ce0bd80,0
I/Dialer: Priority used: 99
MSAudio MSTicker priority increased to maximum.
I/Dialer: Contact has been fixed using proxy
[sip:9010@10.10.149.51] calling [sip:9123@10.10.149.51] on op [0x6fe1c52a00]
Skipping top route of initial route-set because same as request-uri
I/Dialer: bellesip_wake_lock_acquire(): Android wake lock [belle-sip transaction(0x6fe1b4bf20)] acquired [ref=0x3416]
bellesip_wake_lock_acquire(): cast long of wakelock 13334
transaction [0x6fe1b4bf20]: starting transaction background task with id=[3416].
Changing [client] [INVITE] transaction [0x6fe1b4bf20], from state [INIT] to [CALLING]
channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [1071] bytes
INVITE sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.ZkZEHzmja;rport
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51
CSeq: 20 INVITE
Call-ID: WlnkqB4uyY
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: 489
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
User-Agent: Unknown (belle-sip/4.3.0)
v=0
o=9010 2423 2439 IN IP4 10.8.0.100
s=Talk
c=IN IP4 10.8.0.100
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
CallSession [0x6fd335f1d8] moving from state State::OutgoingInit to State::OutgoingProgress

I/Dialer: class mil.navy.nosis.dialer.services.LinphoneServiceTEST LOG I
Linphone core [0x6f7cc45c00] notified [call_state_changed]
E/: error happened here
D/: in onLayout changed
I/Dialer: channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [1071] bytes
INVITE sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.ZkZEHzmja;rport
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51
CSeq: 20 INVITE
Call-ID: WlnkqB4uyY
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: 489
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
User-Agent: Unknown (belle-sip/4.3.0)
v=0
o=9010 2423 2439 IN IP4 10.8.0.100
s=Talk
c=IN IP4 10.8.0.100
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
I/Dialer: channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [1071] bytes
INVITE sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.ZkZEHzmja;rport
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51
CSeq: 20 INVITE
Call-ID: WlnkqB4uyY
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: 489
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
User-Agent: Unknown (belle-sip/4.3.0)
v=0
o=9010 2423 2439 IN IP4 10.8.0.100
s=Talk
c=IN IP4 10.8.0.100
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
I/Nosis Dialer: channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [1071] bytes
INVITE sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.ZkZEHzmja;rport
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51
CSeq: 20 INVITE
Call-ID: WlnkqB4uyY
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: 489
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
User-Agent: Unknown (belle-sip/4.3.0)
v=0
o=9010 2423 2439 IN IP4 10.8.0.100
s=Talk
c=IN IP4 10.8.0.100
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
I/Dialer: channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [1071] bytes
I/Dialer: INVITE sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.ZkZEHzmja;rport
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51
CSeq: 20 INVITE
Call-ID: WlnkqB4uyY
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: 489
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
User-Agent: Unknown (belle-sip/4.3.0)
v=0
o=9010 2423 2439 IN IP4 10.8.0.100
s=Talk
c=IN IP4 10.8.0.100
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
I/Dialer: channel [0x6f7cd34380]: keep alive sent to [UDP://10.10.149.51:5060]
I/Dialer: channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [1071] bytes
INVITE sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.ZkZEHzmja;rport
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51
CSeq: 20 INVITE
Call-ID: WlnkqB4uyY
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: 489
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
User-Agent: Unknown (belle-sip/4.3.0)
v=0
o=9010 2423 2439 IN IP4 10.8.0.100
s=Talk
c=IN IP4 10.8.0.100
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

I/Dialer: channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [1071] bytes
INVITE sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.ZkZEHzmja;rport
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51
CSeq: 20 INVITE
Call-ID: WlnkqB4uyY
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: 489
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
User-Agent: Unknown (belle-sip/4.3.0)
v=0
o=9010 2423 2439 IN IP4 10.8.0.100
s=Talk
c=IN IP4 10.8.0.100
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
I/Dialer: bellesip_wake_lock_acquire(): Android wake lock [belle-sip recv channel] acquired [ref=0x2b26]
bellesip_wake_lock_acquire(): cast long of wakelock 11046
channel [0x6f7cd34380]: starting recv background task with id=[2b26].
channel [0x6f7cd34380]: received [449] new bytes from [UDP://10.10.149.51:5060]:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.0.100:51485;rport=51485;received=10.8.0.100;branch=z9hG4bK.ZkZEHzmja
Call-ID: WlnkqB4uyY
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51;tag=z9hG4bK.ZkZEHzmja
CSeq: 20 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1605118863/5ca617ae6a6ad90e42f09b8cdd215da2”,opaque=“1e48b67962581ba2”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 17.4.0
Content-Length: 0
I/Dialer: channel [0x6f7cd34380] [449] bytes parsed
Found transaction matching response.
Changing [client] [INVITE] transaction [0x6fe1b4bf20], from state [CALLING] to [PROCEEDING]
Changing [client] [INVITE] transaction [0x6fe1b4bf20], from state [PROCEEDING] to [COMPLETED]
I/Dialer: channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [381] bytes
ACK sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.ZkZEHzmja;rport
Call-ID: WlnkqB4uyY
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51;tag=z9hG4bK.ZkZEHzmja
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
Max-Forwards: 70
CSeq: 20 ACK
linphone_core_find_auth_info(): returning auth info username=9010, realm=asterisk
AuthStack::authFound() for Username[9010];Userid;Realm[asterisk];Domain[10.10.149.51];Algorithm;
Auth info found for [9010] realm [asterisk]
I/Dialer: bellesip_wake_lock_acquire(): Android wake lock [belle-sip transaction(0x6fd32e1000)] acquired [ref=0x342a]
bellesip_wake_lock_acquire(): cast long of wakelock 13354
transaction [0x6fd32e1000]: starting transaction background task with id=[342a].
Changing [client] [INVITE] transaction [0x6fd32e1000], from state [INIT] to [CALLING]
channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [1349] bytes
INVITE sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.wsC6IjCnd;rport
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51
CSeq: 21 INVITE
Call-ID: WlnkqB4uyY
I/Dialer: 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: 489
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
User-Agent: Unknown (belle-sip/4.3.0)
Authorization: Digest realm=“asterisk”, nonce=“1605118863/5ca617ae6a6ad90e42f09b8cdd215da2”, algorithm=md5, opaque=“1e48b67962581ba2”, username=“9010”, uri="sip:9123@10.10.149.51", response=“23f6413576c8f42b32c589b922505a89”, cnonce=“PWZKpDvmtuLEbvkS”, nc=00000001, qop=auth
v=0
o=9010 2423 2439 IN IP4 10.8.0.100
s=Talk
c=IN IP4 10.8.0.100
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
channel [0x6f7cd34380]: ending recv background task with id=[2b26].
I/Dialer: bellesip_wake_lock_release(): Android wake lock released [ref=0x2b26]
I/Dialer: bellesip_wake_lock_acquire(): Android wake lock [belle-sip recv channel] acquired [ref=0x3432]
bellesip_wake_lock_acquire(): cast long of wakelock 13362
channel [0x6f7cd34380]: starting recv background task with id=[3432].
channel [0x6f7cd34380]: received [275] new bytes from [UDP://10.10.149.51:5060]:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.0.100:51485;rport=51485;received=10.8.0.100;branch=z9hG4bK.wsC6IjCnd
Call-ID: WlnkqB4uyY
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51
CSeq: 21 INVITE
Server: Asterisk PBX 17.4.0
Content-Length: 0
I/Dialer: channel [0x6f7cd34380] [275] bytes parsed
Found transaction matching response.
Changing [client] [INVITE] transaction [0x6fd32e1000], from state [CALLING] to [PROCEEDING]
op [0x6fe1c52a00] : set_or_update_dialog() current=[0x0] new=[0x0]
Op [0x6fe1c52a00] receiving call response [100], dialog is [0x0] in state [BELLE_SIP_DIALOG_NULL]
channel [0x6f7cd34380]: ending recv background task with id=[3432].
I/Dialer: bellesip_wake_lock_release(): Android wake lock released [ref=0x3432]
I/Dialer: bellesip_wake_lock_acquire(): Android wake lock [belle-sip recv channel] acquired [ref=0x3442]
bellesip_wake_lock_acquire(): cast long of wakelock 13378
channel [0x6f7cd34380]: starting recv background task with id=[3442].
channel [0x6f7cd34380]: received [329] new bytes from [UDP://10.10.149.51:5060]:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 10.8.0.100:51485;rport=51485;received=10.8.0.100;branch=z9hG4bK.wsC6IjCnd
Call-ID: WlnkqB4uyY
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51;tag=7e0836a9-5c51-4292-a8b2-318b9c13499f
CSeq: 21 INVITE
Server: Asterisk PBX 17.4.0
Content-Length: 0
I/Dialer: channel [0x6f7cd34380] [329] bytes parsed
Found transaction matching response.
Changing [client] [INVITE] transaction [0x6fd32e1000], from state [PROCEEDING] to [COMPLETED]
channel [0x6f7cd34380]: message sent to [UDP://10.10.149.51:5060], size: [400] bytes
ACK sip:9123@10.10.149.51 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.100:51485;branch=z9hG4bK.wsC6IjCnd;rport
Call-ID: WlnkqB4uyY
From: sip:9010@10.10.149.51;tag=Q83fF38Zh
To: sip:9123@10.10.149.51;tag=7e0836a9-5c51-4292-a8b2-318b9c13499f
Contact: sip:9010@10.8.0.100:51485;transport=udp;expires=3599;+sip.instance=“urn:uuid:00189c4e-ba22-00e7-8c10-f4666229aaf7”
Max-Forwards: 70
CSeq: 21 ACK
I/Nosis Dialer: op [0x6fe1c52a00] : set_or_update_dialog() current=[0x0] new=[0x0]
Op [0x6fe1c52a00] receiving call response [488], dialog is [0x0] in state [BELLE_SIP_DIALOG_NULL]
Outgoing CallSession [0x6fd335f1d8] failed with SRTP and/or AVPF enabled
CallSession [0x6fd335f1d8] moving from state State::OutgoingProgress to State::Error
Writing echo canceler state, 0 bytes
W/Nosis Dialer: .linphone.ecstate has not been created because there is no data to write
I/Nosis Dialer: ms_filter_unlink: MSRtpRecv:0x6f7cf95900,0–>MSVoidSink:0x6f7ce0bd80,0
=================================================================================
FILTER USAGE STATISTICS
Name Count Time/tick (ms) CPU Usage
min mean max sd
---------------------------------------------------------------------------------
MSRtpRecv 3164 0.01 0.07 0.13 0.02 100.0
MSVoidSink 0 0.00 0.00 0.00 0.00 0.0
MSWebRTCAECM 0 0.00 0.00 0.00 0.00 0.0
MSRtpSend 0 0.00 0.00 0.00 0.00 0.0
=================================================================================
=================================================================================
FILTER USAGE STATISTICS
Name Count Time/tick (ms) CPU Usage
min mean max sd
---------------------------------------------------------------------------------
MSRtpRecv 3164 0.01 0.07 0.13 0.02 100.0
MSVoidSink 0 0.00 0.00 0.00 0.00 0.0
I/Nosis Dialer: MSWebRTCAECM 0 0.00 0.00 0.00 0.00 0.0
MSRtpSend 0 0.00 0.00 0.00 0.00 0.0
=================================================================================
Notifying soundcard that we don’t need it anymore for calls
Stopping ZRTP context on session [0x0]
I/Dialer: ZRTP context destroyed
I/Dialer: MSAudio MSTicker thread exiting
I/Dialer: Linphone core [0x6f7cc45c00] notified [call_log_updated]
Resetting the current call

2

try with the following

disallow=all
allow=ulaw
allow=alaw

3

did not work. l am still getting 488. And l check the endpoint, the endpoint only has ulaw and alaw. l will try add g722 and see if that does anything

4

that did not work… and the log does not have anything useful either

5

It’s very difficult for us to interpret logs from the other system. People know their way around Asterisk logs, so you nee to provide those.

I get the feeling that your logs are only showing one side. Otherwise ACK seems to be being sent to an intermediate response, which is not possible.

6

this is from asterisk console
debuginfo[1].txt (84.8 KB)

this is from var/log/asterisk/messages
messages.txt (7.7 KB)

7
8

l found the problem.


ln my wire shark, l did observer RTP/AVP instead of RTP/SAVP over INVITE request.
l cancelled media encryption on the endpoint, and the problem went away.
Yes, l know that is not the solution, but that is the answer to this problem. l will open separate topic if needed, but that probably involves the app l am building not asterisk
9

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