Inbound calls dropping around 5 mins

Hey everyone, i’m looking for some help here… I have inbound calls dropping almost always around the 4-5 min mark… and can’t quite put my finger on why…

I’ve gathered some logs here to try and isolate the issue… but not sure i’m looking at the right things… outbound calls seem ok.

Here’s from my sip.conf

[TRUNK]
type=peer
port=5060
nat=auto
insecure=port,invite
host=host.com
username=username
fromuser=username
fromdomain=host.com
secret=password
context=from-trunk
canreinvite=no
directmedia=no
disallow=all
allow=ulaw
sendrpid=yes

[inbound]
disallow=all
type=peer
port=5060
nat=auto
insecure=invite
host=host.com
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
directmedia=no
allow=ulaw


[friends_internal](!)
type=peer
host=dynamic
context=from-internal
allow=all
;canreinvite=no
directmedia=no
busylevel=1
allowsubscribe=yes

<— SIP read from UDP:192.168.128.103:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.128.7:5060;branch=z9hG4bK2a07668a
From: “+13215551212” sip:+13215551212@192.168.128.7;tag=as7d7c0ee6
To: sip:3@192.168.128.103:5060;tag=1106770443
Call-ID: 77c6c19f3ec7a30263913f8f0f862350@192.168.128.7:5060
CSeq: 102 INVITE
Contact: sip:3@192.168.128.103:5060
Supported: replaces, path, timer
User-Agent: Grandstream GXP2170 1.0.9.135
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 440

v=0
o=3 8000 8000 IN IP4 192.168.128.103
s=SIP Call
c=IN IP4 192.168.128.103
t=0 0
m=audio 5046 RTP/AVP 0 8 4 18 9 97 111 107 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Jun 16 11:59:59] VERBOSE[1172] chan_sip.c: — (12 headers 20 lines) —
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found RTP audio format 0
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found RTP audio format 8
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found RTP audio format 4
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found RTP audio format 18
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found RTP audio format 9
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found RTP audio format 97
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found RTP audio format 111
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found RTP audio format 107
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found RTP audio format 101
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found audio description format PCMU for ID 0
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found audio description format PCMA for ID 8
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found audio description format G723 for ID 4
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found audio description format G729 for ID 18
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found audio description format G722 for ID 9
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found audio description format iLBC for ID 97
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found audio description format G726-32 for ID 111
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found audio description format opus for ID 107
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Found audio description format telephone-event for ID 101
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263|codec2|g723|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722$
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Peer audio RTP is at port 192.168.128.103:5046
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] sip/route.c: sip_route_dump: route/path hop: sip:3@192.168.128.103:5060
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: set_destination: Parsing sip:3@192.168.128.103:5060 for address/port to send to
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: set_destination: set destination to 192.168.128.103:5060
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Transmitting (no NAT) to 192.168.128.103:5060:
ACK sip:3@192.168.128.103:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.7:5060;branch=z9hG4bK01852c52
Max-Forwards: 70
From: “+13215551212” sip:+13215551212@192.168.128.7;tag=as7d7c0ee6
To: sip:3@192.168.128.103:5060;tag=1106770443
Contact: sip:+13215551212@192.168.128.7:5060
Call-ID: 77c6c19f3ec7a30263913f8f0f862350@192.168.128.7:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.8.0
Content-Length: 0


[Jun 16 11:59:59] VERBOSE[6376][C-00000953] chan_sip.c: Scheduling destruction of SIP dialog ‘2a33d6f86a8afd1b441be9c24ad2d1fb@192.168.128.7:5060’ in 32000 ms (Method: INVITE)
[Jun 16 11:59:59] VERBOSE[6376][C-00000953] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.128.101:5060:
CANCEL sip:1@192.168.128.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.7:5060;branch=z9hG4bK0ebfc308
Max-Forwards: 70
From: “+13215551212” sip:+13215551212@192.168.128.7;tag=as2c7781eb
To: sip:1@192.168.128.101:5060
Call-ID: 2a33d6f86a8afd1b441be9c24ad2d1fb@192.168.128.7:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 16.8.0
Reason: SIP;cause=200;text=“Call completed elsewhere”
Content-Length: 0


[Jun 16 11:59:59] VERBOSE[6376][C-00000953] chan_sip.c: Scheduling destruction of SIP dialog ‘2a33d6f86a8afd1b441be9c24ad2d1fb@192.168.128.7:5060’ in 32000 ms (Method: INVITE)
[Jun 16 11:59:59] VERBOSE[6376][C-00000953] chan_sip.c: Scheduling destruction of SIP dialog ‘49753783410de95b4a463ff276a808b9@192.168.128.7:5060’ in 32000 ms (Method: INVITE)
[Jun 16 11:59:59] VERBOSE[6376][C-00000953] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.128.102:5060:
CANCEL sip:2@192.168.128.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.7:5060;branch=z9hG4bK13ebf32a
Max-Forwards: 70
From: “+13215551212” sip:+13215551212@192.168.128.7;tag=as32f7c198
To: sip:2@192.168.128.102:5060
Call-ID: 49753783410de95b4a463ff276a808b9@192.168.128.7:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 16.8.0
Reason: SIP;cause=200;text=“Call completed elsewhere”
Content-Length: 0


[Jun 16 11:59:59] VERBOSE[6376][C-00000953] chan_sip.c: Scheduling destruction of SIP dialog ‘49753783410de95b4a463ff276a808b9@192.168.128.7:5060’ in 32000 ms (Method: INVITE)
[Jun 16 11:59:59] VERBOSE[6376][C-00000953] chan_sip.c: Scheduling destruction of SIP dialog ‘30d5f4864c77a78264e95d845e99cb69@192.168.128.7:5060’ in 32000 ms (Method: INVITE)
[Jun 16 11:59:59] VERBOSE[6376][C-00000953] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.128.105:5060:
CANCEL sip:5@192.168.128.105:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.7:5060;branch=z9hG4bK7d8f5ad8
Max-Forwards: 70
From: “+13215551212” sip:+13215551212@192.168.128.7;tag=as4df9c419
To: sip:5@192.168.128.105:5060
Call-ID: 30d5f4864c77a78264e95d845e99cb69@192.168.128.7:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 16.8.0
Reason: SIP;cause=200;text=“Call completed elsewhere”
Content-Length: 0


[Jun 16 11:59:59] VERBOSE[6376][C-00000953] chan_sip.c: Scheduling destruction of SIP dialog ‘30d5f4864c77a78264e95d845e99cb69@192.168.128.7:5060’ in 32000 ms (Method: INVITE)
[Jun 16 11:59:59] VERBOSE[1172] chan_sip.c:
<— SIP read from UDP:192.168.128.105:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.128.7:5060;branch=z9hG4bK7d8f5ad8
From: “+13215551212” sip:+13215551212@192.168.128.7;tag=as4df9c419
To: sip:5@192.168.128.105:5060;tag=996741267
Call-ID: 30d5f4864c77a78264e95d845e99cb69@192.168.128.7:5060
CSeq: 102 CANCEL
Contact: sip:5@192.168.128.105:5060
Supported: replaces, path, timer
User-Agent: Grandstream GXP2170 1.0.9.135
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
[Jun 16 11:59:59] VERBOSE[1172] chan_sip.c: — (11 headers 0 lines) —
[Jun 16 11:59:59] VERBOSE[1172] chan_sip.c:
<— SIP read from UDP:192.168.128.105:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.128.7:5060;branch=z9hG4bK7d8f5ad8
From: “+13215551212” sip:+13215551212@192.168.128.7;tag=as4df9c419
To: sip:5@192.168.128.105:5060;tag=996741267
Call-ID: 30d5f4864c77a78264e95d845e99cb69@192.168.128.7:5060
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2170 1.0.9.135
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
[Jun 16 11:59:59] VERBOSE[1172] chan_sip.c: — (10 headers 0 lines) —
[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Transmitting (no NAT) to 192.168.128.105:5060:
ACK sip:5@192.168.128.105:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.7:5060;branch=z9hG4bK7d8f5ad8
Max-Forwards: 70
From: “+13215551212” sip:+13215551212@192.168.128.7;tag=as4df9c419
To: sip:5@192.168.128.105:5060;tag=996741267
Contact: sip:+13215551212@192.168.128.7:5060
Call-ID: 30d5f4864c77a78264e95d845e99cb69@192.168.128.7:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.8.0
Content-Length: 0


[Jun 16 11:59:59] VERBOSE[1172][C-00000953] chan_sip.c: Scheduling destruction of SIP dialog ‘30d5f4864c77a78264e95d845e99cb69@192.168.128.7:5060’ in 32000 ms (Method: INVITE)
[Jun 16 11:59:59] VERBOSE[1172] chan_sip.c:
<— SIP read from UDP:192.168.128.101:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.128.7:5060;branch=z9hG4bK0ebfc308
From: “+13215551212” sip:+13215551212@192.168.128.7;tag=as2c7781eb
To: sip:1@192.168.128.101:5060;tag=19526678
Call-ID: 2a33d6f86a8afd1b441be9c24ad2d1fb@192.168.128.7:5060
CSeq: 102 CANCEL
Contact: sip:1@192.168.128.101:5060
Supported: replaces, path, timer
User-Agent: Grandstream GXP2170 1.0.9.135
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
[Jun 16 11:59:59] VERBOSE[1172] chan_sip.c: — (11 headers 0 lines) —
[Jun 16 11:59:59] VERBOSE[1172] chan_sip.c:
<— SIP read from UDP:192.168.128.102:5060 —>

The log doesn’t show a period of 5 minutes, and, although incomplete, doesn’t show any established call being terminated.

It appears to show a call, dialed to multiple destinations, being connected, and the outgoing calls to the destinations that didn’t win being aborted before being set up. It starts about the time the call connects.

Note canreinvite is an obsolete alias for directmedia, so having both is redundant, given the settings don’t conflict. Also having an inbound and outbound for the same peer is generally not needed, but having inconsistent insecure=port settings for them doesn’t make sense.

I had a look at the code that parse nat=, recently, and there doesn’t seem to be an auto option, although the default is auto_comedia, auto_force_rport.

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