Hi!
Asterisk 13.3.2
I have following trouble: Asterisk realtime with nameb pickup groups and pickup set in features.conf to *8
namedpickupgroup is set
namedcallgroup is set
Phones are in theese groups.
And everything works just fine for some time, but after some time people start to complain, that they can’t pickup any incoming calls, both local and remote.
Here is debug from sip phone trying to pickup incoming call:
SIP Debugging Enabled for IP: 192.168.31.100
[2016-11-29 16:26:30] WARNING[5571][C-000006ab]: channel.c:4611 ast_indicate_data: Unable to handle indication 3 for 'SIP/POD-MSK-000005df'
[2016-11-29 16:26:34] NOTICE[27842]: chan_sip.c:29303 sip_poke_noanswer: Peer '0405' is now UNREACHABLE! Last qualify: 6
<--- SIP read from UDP:192.168.31.100:5060 --->
INVITE sip:*8@asterisk.ariel.loc;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.31.100:5060;branch=z9hG4bK862f67be8fc661084c2651d7b90ac036;rport
From: <sip:0921@asterisk.ariel.loc>;tag=4089273227
To: <sip:*8@asterisk.ariel.loc;user=phone>
Call-ID: 3574977897@192_168_31_100
CSeq: 2 INVITE
Contact: <sip:0921@192.168.31.100:5060>
Max-Forwards: 70
User-Agent: A510 IP/42.207.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER
Content-Type: application/sdp
Content-Length: 385
v=0
o=0921 5014 40 IN IP4 192.168.31.100
s=Mapping
c=IN IP4 192.168.31.100
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
<------------->
--- (14 headers 17 lines) ---
Sending to 192.168.31.100:5060 (no NAT)
Sending to 192.168.31.100:5060 (no NAT)
Using INVITE request as basis request - 3574977897@192_168_31_100
Found peer '0921' for '0921' from 192.168.31.100:5060
<--- Reliably Transmitting (NAT) to 192.168.31.100:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.31.100:5060;branch=z9hG4bK862f67be8fc661084c2651d7b90ac036;received=192.168.31.100;rport=5060
From: <sip:0921@asterisk.ariel.loc>;tag=4089273227
To: <sip:*8@asterisk.ariel.loc;user=phone>;tag=as17ffdbcd
Call-ID: 3574977897@192_168_31_100
CSeq: 2 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c71e419"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '3574977897@192_168_31_100' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.31.100:5060 --->
ACK sip:*8@asterisk.ariel.loc;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.31.100:5060;branch=z9hG4bK862f67be8fc661084c2651d7b90ac036;rport
From: <sip:0921@asterisk.ariel.loc>;tag=4089273227
To: <sip:*8@asterisk.ariel.loc;user=phone>;tag=as17ffdbcd
Call-ID: 3574977897@192_168_31_100
CSeq: 2 ACK
Contact: <sip:0921@192.168.31.100:5060>
Max-Forwards: 70
User-Agent: A510 IP/42.207.00.000.000
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.31.100:5060 --->
INVITE sip:*8@asterisk.ariel.loc;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.31.100:5060;branch=z9hG4bKc62ba18da9daa691c27aa0c4efd58b31;rport
From: <sip:0921@asterisk.ariel.loc>;tag=4089273227
To: <sip:*8@asterisk.ariel.loc;user=phone>
Call-ID: 3574977897@192_168_31_100
CSeq: 3 INVITE
Contact: <sip:0921@192.168.31.100:5060>
Authorization: Digest username="0921", realm="asterisk", algorithm=MD5, uri="sip:*8@asterisk.ariel.loc;user=phone", nonce="3c71e419", response="4c443364d05f5d96fe894ffa5693339e"
Max-Forwards: 70
User-Agent: A510 IP/42.207.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER
Content-Type: application/sdp
Content-Length: 385
v=0
o=0921 5014 40 IN IP4 192.168.31.100
s=Mapping
c=IN IP4 192.168.31.100
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
<------------->
--- (15 headers 17 lines) ---
Sending to 192.168.31.100:5060 (NAT)
Using INVITE request as basis request - 3574977897@192_168_31_100
Found peer '0921' for '0921' from 192.168.31.100:5060
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 96
Found audio description format AAL2-G726-32 for ID 97
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|g726|alaw|g722|g729|g726aal2)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.31.100:5014
Looking for *8 in podolsk-metal (domain asterisk.ariel.loc)
sip_route_dump: route/path hop: <sip:0921@192.168.31.100:5060>
<--- Transmitting (NAT) to 192.168.31.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.31.100:5060;branch=z9hG4bKc62ba18da9daa691c27aa0c4efd58b31;received=192.168.31.100;rport=5060
From: <sip:0921@asterisk.ariel.loc>;tag=4089273227
To: <sip:*8@asterisk.ariel.loc;user=phone>
Call-ID: 3574977897@192_168_31_100
CSeq: 3 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:*8@192.168.70.110:5060>
Content-Length: 0
<------------>
Audio is at 12034
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.31.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.31.100:5060;branch=z9hG4bKc62ba18da9daa691c27aa0c4efd58b31;received=192.168.31.100;rport=5060
From: <sip:0921@asterisk.ariel.loc>;tag=4089273227
To: <sip:*8@asterisk.ariel.loc;user=phone>;tag=as7af14609
Call-ID: 3574977897@192_168_31_100
CSeq: 3 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:*8@192.168.70.110:5060>
Content-Type: application/sdp
Content-Length: 273
v=0
o=Hemul 810626462 810626462 IN IP4 192.168.70.110
s=PBX
c=IN IP4 192.168.70.110
t=0 0
m=audio 12034 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 192.168.31.100:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.31.100:5060;branch=z9hG4bKc62ba18da9daa691c27aa0c4efd58b31;received=192.168.31.100;rport=5060
From: <sip:0921@asterisk.ariel.loc>;tag=4089273227
To: <sip:*8@asterisk.ariel.loc;user=phone>;tag=as7af14609
Call-ID: 3574977897@192_168_31_100
CSeq: 3 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:*8@192.168.70.110:5060>
Content-Type: application/sdp
Content-Length: 273
v=0
o=Hemul 810626462 810626462 IN IP4 192.168.70.110
s=PBX
c=IN IP4 192.168.70.110
t=0 0
m=audio 12034 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:192.168.31.100:5060 --->
ACK sip:*8@192.168.70.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.31.100:5060;branch=z9hG4bKb7b4152eb4fec8962b3362557bab9ffc;rport
From: <sip:0921@asterisk.ariel.loc>;tag=4089273227
To: <sip:*8@asterisk.ariel.loc;user=phone>;tag=as7af14609
Call-ID: 3574977897@192_168_31_100
CSeq: 3 ACK
Contact: <sip:0921@192.168.31.100:5060>
Authorization: Digest username="0921", realm="asterisk", algorithm=MD5, uri="sip:*8@asterisk.ariel.loc;user=phone", nonce="3c71e419", response="4c443364d05f5d96fe894ffa5693339e"
Max-Forwards: 70
User-Agent: A510 IP/42.207.00.000.000
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:192.168.31.100:5060 --->
<------------->
<--- SIP read from UDP:192.168.31.100:5060 --->
ACK sip:*8@192.168.70.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.31.100:5060;branch=z9hG4bKc06d4a8ce2d4cd767db5904619f1a1d3;rport
From: <sip:0921@asterisk.ariel.loc>;tag=4089273227
To: <sip:*8@asterisk.ariel.loc;user=phone>;tag=as7af14609
Call-ID: 3574977897@192_168_31_100
CSeq: 3 ACK
Contact: <sip:0921@192.168.31.100:5060>
Authorization: Digest username="0921", realm="asterisk", algorithm=MD5, uri="sip:*8@asterisk.ariel.loc;user=phone", nonce="3c71e419", response="4c443364d05f5d96fe894ffa5693339e"
Max-Forwards: 70
User-Agent: A510 IP/42.207.00.000.000
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Scheduling destruction of SIP dialog '3574977897@192_168_31_100' in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 192.168.31.100:5060:
BYE sip:0921@192.168.31.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.70.110:5060;branch=z9hG4bK16b0ef65;rport
Max-Forwards: 70
From: <sip:*8@asterisk.ariel.loc;user=phone>;tag=as7af14609
To: <sip:0921@asterisk.ariel.loc>;tag=4089273227
Call-ID: 3574977897@192_168_31_100
CSeq: 102 BYE
User-Agent: PBX
Proxy-Authorization: Digest username="0921", realm="asterisk", algorithm=MD5, uri="sip:asterisk.ariel.loc", nonce="3c71e419", response="a46dd10fd1a0ebe09fd00f002291e50f"
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0
---
<--- SIP read from UDP:192.168.31.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.70.110:5060;branch=z9hG4bK16b0ef65;rport=5060
From: <sip:*8@asterisk.ariel.loc;user=phone>;tag=as7af14609
To: <sip:0921@asterisk.ariel.loc>;tag=4089273227
Call-ID: 3574977897@192_168_31_100
CSeq: 102 BYE
Contact: <sip:0921@192.168.31.100:5060>
Supported: replaces
User-Agent: A510 IP/42.207.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '3574977897@192_168_31_100' Method: ACK