'481 Call Leg/Transaction Does Not Exist' from Gigaset N510 IP Pro

Asterisk Asterisk SIP
1

Hi,

I’m about to migrate an old Asterisk system with chan_sip to the newest version with PJSIP.

I have a problem with a Gigaset N510 IP Pro DECT station. In some repeatable scenarios, Asterisk can’t end the call by sending “BYE”, because the phone answers with 481. The same phone works flawlessly with the old Asterisk version with chan_sip, and it also works on the new system if called by another phone.

I had a look at the SIP traces, but frankly speaking I’m not able to see the issue.

Maybe someone could have a look and see if there is something obvious?

<--- Received SIP request (982 bytes) from UDP:192.168.1.115:5060 --->
INVITE sip:999@pbx.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bKc184c6c4d2fc7a06e346fe3ec56101f7;rport
From:  <sip:111@pbx.lan>;tag=267741813
To: <sip:999@pbx.lan;user=phone>
Call-ID: 1332342870@192_168_1_115
CSeq: 2 INVITE
Contact: <sip:111@192.168.1.115:5060>
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381

v=0
o=111 5014 6 IN IP4 192.168.1.115
s=Mapping
c=IN IP4 192.168.1.115
t=0 0
m=audio 5014 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


<--- Transmitting SIP response (521 bytes) to UDP:192.168.1.115:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.115:5060;rport=5060;received=192.168.1.115;branch=z9hG4bKc184c6c4d2fc7a06e346fe3ec56101f7
Call-ID: 1332342870@192_168_1_115
From: <sip:111@pbx.lan>;tag=267741813
To: <sip:999@pbx.lan;user=phone>;tag=z9hG4bKc184c6c4d2fc7a06e346fe3ec56101f7
CSeq: 2 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1615291956/a51f580c40e27fd65e532841e26e72e8",opaque="34d5617b3babc562",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.2.2
Content-Length:  0


<--- Received SIP request (435 bytes) from UDP:192.168.1.115:5060 --->
ACK sip:999@pbx.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bKc184c6c4d2fc7a06e346fe3ec56101f7;rport
From:  <sip:111@pbx.lan>;tag=267741813
To: <sip:999@pbx.lan;user=phone>;tag=z9hG4bKc184c6c4d2fc7a06e346fe3ec56101f7
Call-ID: 1332342870@192_168_1_115
CSeq: 2 ACK
Contact: <sip:111@192.168.1.115:5060>
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Content-Length: 0


<--- Received SIP request (1288 bytes) from UDP:192.168.1.115:5060 --->
INVITE sip:999@pbx.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK51915be5eda3b8fa2070c0c97bc4a9c3;rport
From:  <sip:111@pbx.lan>;tag=267741813
To: <sip:999@pbx.lan;user=phone>
Call-ID: 1332342870@192_168_1_115
CSeq: 3 INVITE
Contact: <sip:111@192.168.1.115:5060>
Authorization: Digest username="user111", realm="asterisk", qop=auth, algorithm=md5, uri="sip:999@pbx.lan;user=phone", nonce="1615291956/a51f580c40e27fd65e532841e26e72e8", nc=00000001, cnonce="bedf5430c84175688e47387f74c66ede", opaque="34d5617b3babc562", response="f8293a7a024d00175c5bd8083559076b"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381

v=0
o=111 5014 6 IN IP4 192.168.1.115
s=Mapping
c=IN IP4 192.168.1.115
t=0 0
m=audio 5014 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


<--- Transmitting SIP response (325 bytes) to UDP:192.168.1.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.115:5060;rport=5060;received=192.168.1.115;branch=z9hG4bK51915be5eda3b8fa2070c0c97bc4a9c3
Call-ID: 1332342870@192_168_1_115
From: <sip:111@pbx.lan>;tag=267741813
To: <sip:999@pbx.lan;user=phone>
CSeq: 3 INVITE
Server: Asterisk PBX 18.2.2
Content-Length:  0


<--- Transmitting SIP response (839 bytes) to UDP:192.168.1.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.115:5060;rport=5060;received=192.168.1.115;branch=z9hG4bK51915be5eda3b8fa2070c0c97bc4a9c3
Call-ID: 1332342870@192_168_1_115
From: <sip:111@pbx.lan>;tag=267741813
To: <sip:999@pbx.lan;user=phone>;tag=b0e9480e-7680-4733-8f04-f1e864439cb8
CSeq: 3 INVITE
Server: Asterisk PBX 18.2.2
Contact: <sip:192.168.1.111:50060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   248

v=0
o=- 5014 8 IN IP4 192.168.1.111
s=Asterisk
c=IN IP4 192.168.1.111
t=0 0
m=audio 10034 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


<--- Received SIP request (729 bytes) from UDP:192.168.1.115:5060 --->
ACK sip:192.168.1.111:50060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK8dab2b0116eeb5c119079254c0589135;rport
From:  <sip:111@pbx.lan>;tag=267741813
To: <sip:999@pbx.lan;user=phone>;tag=b0e9480e-7680-4733-8f04-f1e864439cb8
Call-ID: 1332342870@192_168_1_115
CSeq: 3 ACK
Contact: <sip:111@192.168.1.115:5060>
Authorization: Digest username="user111", realm="asterisk", qop=auth, algorithm=md5, uri="sip:999@pbx.lan;user=phone", nonce="1615291956/a51f580c40e27fd65e532841e26e72e8", nc=00000001, cnonce="bedf5430c84175688e47387f74c66ede", opaque="34d5617b3babc562", response="f8293a7a024d00175c5bd8083559076b"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Content-Length: 0


<--- Transmitting SIP request (412 bytes) to UDP:192.168.1.115:5060 --->
BYE sip:111@192.168.1.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:50060;rport;branch=z9hG4bKPj44f3d6f3-cbe7-40a0-bcb9-faa806367525
From: <sip:999@pbx.lan;user=phone>;tag=b0e9480e-7680-4733-8f04-f1e864439cb8
To: <sip:111@pbx.lan>;tag=267741813
Call-ID: 1332342870@192_168_1_115
CSeq: 13492 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 18.2.2
Content-Length:  0


<--- Received SIP response (394 bytes) from UDP:192.168.1.115:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.1.111:50060;rport=50060;branch=z9hG4bKPj44f3d6f3-cbe7-40a0-bcb9-faa806367525
From: <sip:999@pbx.lan;user=phone>;tag=b0e9480e-7680-4733-8f04-f1e864439cb8
To: <sip:111@pbx.lan>;tag=267741813
Call-ID: 1332342870@192_168_1_115
CSeq: 13492 BYE
User-Agent: N510 IP PRO/42.258.00.000.000
Content-Length: 0

Firmware of the Gigaset base is up-to-date.

2

From the perspective of the SIP signaling I don’t see any reason for the Gigaset base to send a 481 to our BYE request. It looks as if it is formed correctly.

Can you provide a working call in the same direction from chan_sip for comparison?

3

Yes, here it is:

<--- SIP read from UDP:192.168.1.115:5060 --->
INVITE sip:999@asterisk.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK2de6a00edc13e16a2f810005c7b95db8;rport
From: <sip:111@asterisk.lan>;tag=292623808
To: <sip:999@asterisk.lan;user=phone>
Call-ID: 2844704814@192_168_1_115
CSeq: 2 INVITE
Contact: <sip:111@192.168.1.115:5060>
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381

v=0
o=111 5006 5 IN IP4 192.168.1.115
s=Mapping
c=IN IP4 192.168.1.115
t=0 0
m=audio 5006 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


<--- Reliably Transmitting (no NAT) to 192.168.1.115:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK2de6a00edc13e16a2f810005c7b95db8;received=192.168.1.115;rport=5060
From: <sip:111@asterisk.lan>;tag=292623808
To: <sip:999@asterisk.lan;user=phone>;tag=as5e55f5ea
Call-ID: 2844704814@192_168_1_115
CSeq: 2 INVITE
Server: Asterisk PBX 10.12.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="55cfb424"
Content-Length: 0


<--- SIP read from UDP:192.168.1.115:5060 --->
ACK sip:999@asterisk.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK2de6a00edc13e16a2f810005c7b95db8;rport
From: <sip:111@asterisk.lan>;tag=292623808
To: <sip:999@asterisk.lan;user=phone>;tag=as5e55f5ea
Call-ID: 2844704814@192_168_1_115
CSeq: 2 ACK
Contact: <sip:111@192.168.1.115:5060>
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Content-Length: 0


<--- SIP read from UDP:192.168.1.115:5060 --->
INVITE sip:999@asterisk.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bKcd2a787186fd02cdbc1efd3ebf61bf;rport
From: <sip:111@asterisk.lan>;tag=292623808
To: <sip:999@asterisk.lan;user=phone>
Call-ID: 2844704814@192_168_1_115
CSeq: 3 INVITE
Contact: <sip:111@192.168.1.115:5060>
Authorization: Digest username="111", realm="asterisk", algorithm=MD5, uri="sip:999@asterisk.lan;user=phone", nonce="55cfb424", response="eaf6efd4f8a7bec6773c52e363d434a3"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381

v=0
o=111 5006 5 IN IP4 192.168.1.115
s=Mapping
c=IN IP4 192.168.1.115
t=0 0
m=audio 5006 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


<--- Transmitting (no NAT) to 192.168.1.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bKcd2a787186fd02cdbc1efd3ebf61bf;received=192.168.1.115;rport=5060
From: <sip:111@asterisk.lan>;tag=292623808
To: <sip:999@asterisk.lan;user=phone>
Call-ID: 2844704814@192_168_1_115
CSeq: 3 INVITE
Server: Asterisk PBX 10.12.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:999@192.168.1.105:50060>
Content-Length: 0


<--- Reliably Transmitting (no NAT) to 192.168.1.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bKcd2a787186fd02cdbc1efd3ebf61bf;received=192.168.1.115;rport=5060
From: <sip:111@asterisk.lan>;tag=292623808
To: <sip:999@asterisk.lan;user=phone>;tag=as3ab9e2e7
Call-ID: 2844704814@192_168_1_115
CSeq: 3 INVITE
Server: Asterisk PBX 10.12.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:999@192.168.1.105:50060>
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 1105605523 1105605523 IN IP4 192.168.1.105
s=Asterisk PBX 10.12.3
c=IN IP4 192.168.1.105
t=0 0
m=audio 10064 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


<--- SIP read from UDP:192.168.1.115:5060 --->
ACK sip:999@192.168.1.105:50060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK3a3cc9cfea390ef270312fe750fa9a21;rport
From: <sip:111@asterisk.lan>;tag=292623808
To: <sip:999@asterisk.lan;user=phone>;tag=as3ab9e2e7
Call-ID: 2844704814@192_168_1_115
CSeq: 3 ACK
Contact: <sip:111@192.168.1.115:5060>
Authorization: Digest username="111", realm="asterisk", algorithm=MD5, uri="sip:999@asterisk.lan;user=phone", nonce="55cfb424", response="eaf6efd4f8a7bec6773c52e363d434a3"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Content-Length: 0


Reliably Transmitting (no NAT) to 192.168.1.115:5060:
BYE sip:111@192.168.1.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:50060;branch=z9hG4bK760c1d92;rport
Max-Forwards: 70
From: <sip:999@asterisk.lan;user=phone>;tag=as3ab9e2e7
To: <sip:111@asterisk.lan>;tag=292623808
Call-ID: 2844704814@192_168_1_115
CSeq: 102 BYE
User-Agent: Asterisk PBX 10.12.3
Proxy-Authorization: Digest username="111", realm="asterisk", algorithm=MD5, uri="sip:asterisk.lan", nonce="", response="1c356d3a947495fa693f3fc2c321d973"
Reason: Q.850;cause=16
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<--- SIP read from UDP:192.168.1.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.105:50060;branch=z9hG4bK760c1d92;rport=50060
From: <sip:999@asterisk.lan;user=phone>;tag=as3ab9e2e7
To: <sip:111@asterisk.lan>;tag=292623808
Call-ID: 2844704814@192_168_1_115
CSeq: 102 BYE
Contact: <sip:111@192.168.1.115:5060>
Supported: replaces
User-Agent: N510 IP PRO/42.258.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Length: 0

It’s the same network, the same DECT station (it supports multiple SIP accounts), and the same dialplan with the exceptions of the modifications needed to migrate from chan_sip to chan_pjsip.

4

The only thing of note I see is that chan_sip in the version of Asterisk you’re using adds a Proxy-Authorization header to the BYE, when it shouldn’t. Otherwise the BYE requests appear correct in both cases.

Does chan_sip work in 18?

5

This is… effort. I will recompile Asterisk 18 with chan_sip support and try. Will take some time.

I’m really looking forward to migrate to the newest Asterisk version and the new hardware :star_struck:

6

Ok, after some fighting: I get a 481 with chan_sip also in Asterisk 18.2.2.

Here is the SIP trace:

<--- SIP read from UDP:192.168.1.115:5060 --->
INVITE sip:999@pbx.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bKb732987f62321b11ce79d4f246d39ae;rport
From: <sip:111@pbx.lan>;tag=781587539
To: <sip:999@pbx.lan;user=phone>
Call-ID: 4144943370@192_168_1_115
CSeq: 2 INVITE
Contact: <sip:111@192.168.1.115:5060>
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381

v=0
o=111 5006 2 IN IP4 192.168.1.115
s=Mapping
c=IN IP4 192.168.1.115
t=0 0
m=audio 5006 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


<--- Reliably Transmitting (no NAT) to 192.168.1.115:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bKb732987f62321b11ce79d4f246d39ae;received=192.168.1.115;rport=5060
From: <sip:111@pbx.lan>;tag=781587539
To: <sip:999@pbx.lan;user=phone>;tag=as355d5068
Call-ID: 4144943370@192_168_1_115
CSeq: 2 INVITE
Server: Asterisk PBX 18.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f054830"
Content-Length: 0


<--- SIP read from UDP:192.168.1.115:5060 --->
ACK sip:999@pbx.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bKb732987f62321b11ce79d4f246d39ae;rport
From: <sip:111@pbx.lan>;tag=781587539
To: <sip:999@pbx.lan;user=phone>;tag=as355d5068
Call-ID: 4144943370@192_168_1_115
CSeq: 2 ACK
Contact: <sip:111@192.168.1.115:5060>
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Content-Length: 0


<--- SIP read from UDP:192.168.1.115:5060 --->
INVITE sip:999@pbx.lan;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK1d27aa2dea8a1c38917f029275764a80;rport
From: <sip:111@pbx.lan>;tag=781587539
To: <sip:999@pbx.lan;user=phone>
Call-ID: 4144943370@192_168_1_115
CSeq: 3 INVITE
Contact: <sip:111@192.168.1.115:5060>
Authorization: Digest username="111", realm="asterisk", algorithm=MD5, uri="sip:999@pbx.lan;user=phone", nonce="2f054830", response="d3ddff40a65f435bb66ce3bde2cf1131"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381

v=0
o=111 5006 2 IN IP4 192.168.1.115
s=Mapping
c=IN IP4 192.168.1.115
t=0 0
m=audio 5006 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


<--- Transmitting (no NAT) to 192.168.1.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK1d27aa2dea8a1c38917f029275764a80;received=192.168.1.115;rport=5060
From: <sip:111@pbx.lan>;tag=781587539
To: <sip:999@pbx.lan;user=phone>
Call-ID: 4144943370@192_168_1_115
CSeq: 3 INVITE
Server: Asterisk PBX 18.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:999@192.168.1.111:50060>
Content-Length: 0


<--- Reliably Transmitting (no NAT) to 192.168.1.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK1d27aa2dea8a1c38917f029275764a80;received=192.168.1.115;rport=5060
From: <sip:111@pbx.lan>;tag=781587539
To: <sip:999@pbx.lan;user=phone>;tag=as3716515e
Call-ID: 4144943370@192_168_1_115
CSeq: 3 INVITE
Server: Asterisk PBX 18.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:999@192.168.1.111:50060>
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1772566906 1772566906 IN IP4 192.168.1.111
s=Asterisk PBX 18.2.2
c=IN IP4 192.168.1.111
t=0 0
m=audio 10068 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


<--- SIP read from UDP:192.168.1.115:5060 --->
ACK sip:999@192.168.1.111:50060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bKd9f68bc3697ea3b2085b4463e11dd7d;rport
From: <sip:111@pbx.lan>;tag=781587539
To: <sip:999@pbx.lan;user=phone>;tag=as3716515e
Call-ID: 4144943370@192_168_1_115
CSeq: 3 ACK
Contact: <sip:111@192.168.1.115:5060>
Authorization: Digest username="111", realm="asterisk", algorithm=MD5, uri="sip:999@pbx.lan;user=phone", nonce="2f054830", response="d3ddff40a65f435bb66ce3bde2cf1131"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.258.00.000.000
Content-Length: 0


Reliably Transmitting (no NAT) to 192.168.1.115:5060:
BYE sip:111@192.168.1.115:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:50060;branch=z9hG4bK3e07e5bf;rport
Max-Forwards: 70
From: <sip:999@pbx.lan;user=phone>;tag=as3716515e
To: <sip:111@pbx.lan>;tag=781587539
Call-ID: 4144943370@192_168_1_115
CSeq: 102 BYE
User-Agent: Asterisk PBX 18.2.2
Proxy-Authorization: Digest username="111", realm="asterisk", algorithm=MD5, uri="sip:pbx.lan", nonce="2f054830", response="21cf2f640a0de03fc137ce8da58e778c"
Reason: Q.850;cause=16
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<--- SIP read from UDP:192.168.1.115:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.1.111:50060;branch=z9hG4bK3e07e5bf;rport=50060
From: <sip:999@pbx.lan;user=phone>;tag=as3716515e
To: <sip:111@pbx.lan>;tag=781587539
Call-ID: 4144943370@192_168_1_115
CSeq: 102 BYE
User-Agent: N510 IP PRO/42.258.00.000.000
Content-Length: 0

A little bit offtopic: Is there an Asterisk internal database / cache / whatever, which could cause configurations to (maybe falsely) survive restarts, even though they were removed from the configuration files? I struggled a considerable amount of time, because the value of a “username” parameter in sip.conf survived several restarts, even though it was removed in sip.conf. I grep’ed the whole Asterisk filesystem tree and verified the value was not there, nevertheless I could see the value with “sip show peer 111” in the “Def. Username” line. Where does it hide from restarts?

7

Ok, I think I got it.

It seems to be a bug in the N510 IP Pro base firmware.

The N510 IP Pro allows up to 6 SIP accounts, initially named IP1 to IP6.

IP1 is configured as the productive account and used ever since without problems.

IP2 is configured for testing purpose on the new system. Here I get the 481 issue.

Since the SIP traces really look flawlessly, all fingers are pointing to the Gigaset. I started to experiment:
If I deactivate IP1 in the webinterface of the Gigaset, magically IP2 starts to work as expected. Seems, that IP2 is somehow falsely validated against IP1 parameters sometimes, if IP1 is active.

Next step: Convince Gigaset to provide a bug fix.

Thanks for the support.

8

So, to finish the story:

I contacted Gigaset, but they are not willing to give support to anyone else than “authorized and certified Gigaset Pro partners”. Unfortunately, I bought the base by a big German reseller, who is not one of these partners.

So, definitively no recommendation for any “Gigaset Pro” product - there is no support. :-1:

9

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