I’m running Asterisk with FreePbx using Grandstream HT-286 ATA’s. For some reason every call hangs up at the 17:28 (1048 secs) mark. This even happens from one extension to another on the same lan with the server.
Any ideas on whats going on?
Not without the sip debug logs.
post a screenshot
-Jake
www.voipcitadel.com
I’m experiencing the same issue.
Here is the SIP DEBUG LOG:
<------------->
[Sep 10 14:42:25] VERBOSE[2318] chan_sip.c: — (9 headers 0 lines) —
[Sep 10 14:42:25] VERBOSE[2318] chan_sip.c: Scheduling destruction of SIP dialog 'XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Sep 10 14:42:25] NOTICE[2318] chan_sip.c: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
[Sep 10 14:42:30] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:1024 —>
<------------->
[Sep 10 14:42:31] VERBOSE[2318] chan_sip.c: Really destroying SIP dialog 'XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com’ Method: REGISTER
[Sep 10 14:42:49] NOTICE[2318] chan_sip.c: – Re-registration for XXXXXXXXXX@sip.broadvoice.com
[Sep 10 14:42:49] VERBOSE[2318] dnsmgr.c: > doing dnsmgr_lookup for ‘sip.broadvoice.com’
[Sep 10 14:42:49] VERBOSE[2318] chan_sip.c: REGISTER 11 headers, 0 lines
[Sep 10 14:42:49] VERBOSE[2318] chan_sip.c: Reliably Transmitting (no NAT) to 206.15.156.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2acece68;rport
Max-Forwards: 70
From: sip:XXXXXXXXXX@sip.broadvoice.com;tag=as031989a6
To: sip:XXXXXXXXXX@sip.broadvoice.com
Call-ID: XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com
CSeq: 778 REGISTER
User-Agent: Asterisk PBX 1.6.2.11
Authorization: Digest username=“XXXXXXXXXX”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXgdxc3l12T6kfyleBW”, response=“6f44f8ad80ddd1a06dbb54310daf4dbb”, qop=auth, cnonce=“752423db”, nc=0000025c
Expires: 120
Contact: sip:XXXXXXXXXX@XXX.XXX.XXX.XXX
Content-Length: 0
[Sep 10 14:42:49] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com
CSeq: 778 REGISTER
From: sip:XXXXXXXXXX@sip.broadvoice.com;tag=as031989a6
To: sip:XXXXXXXXXX@sip.broadvoice.com
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2acece68;rport=5060
Contact: sip:XXXXXXXXXX@XXX.XXX.XXX.XXX
Expires: 30
Content-Length: 0
<------------->
[Sep 10 14:42:49] VERBOSE[2318] chan_sip.c: — (9 headers 0 lines) —
[Sep 10 14:42:49] VERBOSE[2318] chan_sip.c: Scheduling destruction of SIP dialog 'XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Sep 10 14:42:49] NOTICE[2318] chan_sip.c: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
[Sep 10 14:42:50] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:1024 —>
<------------->
[Sep 10 14:42:55] VERBOSE[2318] chan_sip.c: Really destroying SIP dialog 'XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com’ Method: REGISTER
[Sep 10 14:42:57] VERBOSE[2318] chan_sip.c: Reliably Transmitting (no NAT) to 206.15.156.221:5060:
OPTIONS sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK7f7b6bc1;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@XXX.XXX.XXX.XXX;tag=as199c7967
To: sip:sip.broadvoice.com
Contact: sip:Unknown@XXX.XXX.XXX.XXX
Call-ID: 36c3618974f5af24041b5fa047002a3a@XXX.XXX.XXX.XXX
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Fri, 10 Sep 2010 21:42:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
[Sep 10 14:42:57] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: 36c3618974f5af24041b5fa047002a3a@XXX.XXX.XXX.XXX
CSeq: 102 OPTIONS
From: “Unknown” sip:Unknown@XXX.XXX.XXX.XXX;tag=as199c7967
To: sip:sip.broadvoice.com
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK7f7b6bc1;rport=5060
Supported: 100rel
Max-Forwards: 70
Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0
<------------->
[Sep 10 14:42:57] VERBOSE[2318] chan_sip.c: — (14 headers 0 lines) —
[Sep 10 14:42:57] VERBOSE[2318] chan_sip.c: Really destroying SIP dialog '36c3618974f5af24041b5fa047002a3a@XXX.XXX.XXX.XXX’ Method: OPTIONS
[Sep 10 14:43:10] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:1024 —>
<------------->
[Sep 10 14:43:13] NOTICE[2318] chan_sip.c: – Re-registration for XXXXXXXXXX@sip.broadvoice.com
[Sep 10 14:43:13] VERBOSE[2318] dnsmgr.c: > doing dnsmgr_lookup for ‘sip.broadvoice.com’
[Sep 10 14:43:13] VERBOSE[2318] chan_sip.c: REGISTER 11 headers, 0 lines
[Sep 10 14:43:13] VERBOSE[2318] chan_sip.c: Reliably Transmitting (no NAT) to 206.15.156.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0ee32803;rport
Max-Forwards: 70
From: sip:XXXXXXXXXX@sip.broadvoice.com;tag=as6bc964fe
To: sip:XXXXXXXXXX@sip.broadvoice.com
Call-ID: XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com
CSeq: 779 REGISTER
User-Agent: Asterisk PBX 1.6.2.11
Authorization: Digest username=“XXXXXXXXXX”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXgdxc3l12T6kfyleBW”, response=“fbbbf2e00d639a6a9576b528cc1a3d5c”, qop=auth, cnonce=“22905980”, nc=0000025d
Expires: 120
Contact: sip:XXXXXXXXXX@XXX.XXX.XXX.XXX
Content-Length: 0
[Sep 10 14:43:13] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com
CSeq: 779 REGISTER
From: sip:XXXXXXXXXX@sip.broadvoice.com;tag=as6bc964fe
To: sip:XXXXXXXXXX@sip.broadvoice.com
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0ee32803;rport=5060
Contact: sip:XXXXXXXXXX@XXX.XXX.XXX.XXX
Expires: 30
Content-Length: 0
<------------->
[Sep 10 14:43:13] VERBOSE[2318] chan_sip.c: — (9 headers 0 lines) —
[Sep 10 14:43:13] VERBOSE[2318] chan_sip.c: Scheduling destruction of SIP dialog 'XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Sep 10 14:43:13] NOTICE[2318] chan_sip.c: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
[Sep 10 14:43:14] VERBOSE[2318] chan_sip.c: Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:1024:
OPTIONS sip:100@172.16.0.124 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK564afc73;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@XXX.XXX.XXX.XXX;tag=as4879a336
To: sip:100@172.16.0.124
Contact: sip:Unknown@XXX.XXX.XXX.XXX
Call-ID: 170df993676949f6469e23681ad27750@XXX.XXX.XXX.XXX
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Fri, 10 Sep 2010 21:43:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
[Sep 10 14:43:14] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:1024 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK564afc73;rport
From: “Unknown” sip:Unknown@XXX.XXX.XXX.XXX;tag=as4879a336
To: sip:100@172.16.0.124;tag=as5db45778
Call-ID: 170df993676949f6469e23681ad27750@XXX.XXX.XXX.XXX
CSeq: 102 OPTIONS
User-Agent: Grandstream HT287 1.1.0.45 DevId 000b822193d2
Session-Expires: 180;refresher=uac
Min-SE: 180
Require: timer
Contact: sip:100@172.16.0.124
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Supported: replaces, timer
Content-Length: 0
<------------->
[Sep 10 14:43:14] VERBOSE[2318] chan_sip.c: — (14 headers 0 lines) —
[Sep 10 14:43:14] VERBOSE[2318] chan_sip.c: Really destroying SIP dialog '170df993676949f6469e23681ad27750@XXX.XXX.XXX.XXX’ Method: OPTIONS
[Sep 10 14:43:19] VERBOSE[2318] chan_sip.c: Really destroying SIP dialog 'XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com’ Method: REGISTER
[Sep 10 14:43:30] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:1024 —>
<------------->
[Sep 10 14:43:37] NOTICE[2318] chan_sip.c: – Re-registration for XXXXXXXXXX@sip.broadvoice.com
[Sep 10 14:43:37] VERBOSE[2318] dnsmgr.c: > doing dnsmgr_lookup for ‘sip.broadvoice.com’
[Sep 10 14:43:37] VERBOSE[2318] chan_sip.c: REGISTER 11 headers, 0 lines
[Sep 10 14:43:37] VERBOSE[2318] chan_sip.c: Reliably Transmitting (no NAT) to 206.15.156.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK1222bf84;rport
Max-Forwards: 70
From: sip:XXXXXXXXXX@sip.broadvoice.com;tag=as1600c3ff
To: sip:XXXXXXXXXX@sip.broadvoice.com
Call-ID: XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com
CSeq: 780 REGISTER
User-Agent: Asterisk PBX 1.6.2.11
Authorization: Digest username=“XXXXXXXXXX”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXgdxc3l12T6kfyleBW”, response=“6ed373188d741a1f0458cbaf722c8535”, qop=auth, cnonce=“2de782f4”, nc=0000025e
Expires: 120
Contact: sip:XXXXXXXXXX@XXX.XXX.XXX.XXX
Content-Length: 0
[Sep 10 14:43:37] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com
CSeq: 780 REGISTER
From: sip:XXXXXXXXXX@sip.broadvoice.com;tag=as1600c3ff
To: sip:XXXXXXXXXX@sip.broadvoice.com
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK1222bf84;rport=5060
Contact: sip:XXXXXXXXXX@XXX.XXX.XXX.XXX
Expires: 30
Content-Length: 0
<------------->
[Sep 10 14:43:37] VERBOSE[2318] chan_sip.c: — (9 headers 0 lines) —
[Sep 10 14:43:37] VERBOSE[2318] chan_sip.c: Scheduling destruction of SIP dialog 'XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Sep 10 14:43:37] NOTICE[2318] chan_sip.c: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c: Really destroying SIP dialog 'XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@sip.broadvoice.com’ Method: REGISTER
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:1024 —>
BYE sip:XXXXXXXXXX@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 172.16.0.124;branch=z9hG4bK5c37586dad4e2fc3
From: “Home” sip:100@voip.XXXXXX.com;tag=779ee87e889e987e
To: sip:XXXXXXXXXX@voip.XXXXXX.com;tag=as5db45778
Supported: replaces, timer
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri="sip:XXXXXXXXXX@XXX.XXX.XXX.XXX", nonce=“230fbcd8”, response="28ad81a7ede50afb37b2436ba5c09a87"
Call-ID: c6ccedb92608044c@172.16.0.124
CSeq: 52466 BYE
User-Agent: Grandstream HT287 1.1.0.45 DevId 000b822193d2
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
<------------->
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c: — (12 headers 0 lines) —
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c: Sending to XXX.XXX.XXX.XXX : 1024 (NAT)
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c:
<— Transmitting (NAT) to XXX.XXX.XXX.XXX:1024 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.124;branch=z9hG4bK5c37586dad4e2fc3;received=XXX.XXX.XXX.XXX
From: “Home” sip:100@voip.XXXXXX.com;tag=779ee87e889e987e
To: sip:XXXXXXXXXX@voip.XXXXXX.com;tag=as5db45778
Call-ID: c6ccedb92608044c@172.16.0.124
CSeq: 52466 BYE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Sep 10 14:43:43] VERBOSE[9792] pbx.c: – Executing [h@macro-dialout-trunk:1] Macro(“SIP/100-00000123”, “hangupcall,”) in new stack
[Sep 10 14:43:43] VERBOSE[9792] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/100-00000123”, “1?skiprg”) in new stack
[Sep 10 14:43:43] VERBOSE[9792] pbx.c: – Goto (macro-hangupcall,s,4)
[Sep 10 14:43:43] VERBOSE[9792] pbx.c: – Executing [s@macro-hangupcall:4] GotoIf(“SIP/100-00000123”, “1?skipblkvm”) in new stack
[Sep 10 14:43:43] VERBOSE[9792] pbx.c: – Goto (macro-hangupcall,s,7)
[Sep 10 14:43:43] VERBOSE[9792] pbx.c: – Executing [s@macro-hangupcall:7] GotoIf(“SIP/100-00000123”, “1?theend”) in new stack
[Sep 10 14:43:43] VERBOSE[9792] pbx.c: – Goto (macro-hangupcall,s,9)
[Sep 10 14:43:43] VERBOSE[9792] pbx.c: – Executing [s@macro-hangupcall:9] Hangup(“SIP/100-00000123”, “”) in new stack
[Sep 10 14:43:43] VERBOSE[9792] app_macro.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/100-00000123’ in macro ‘hangupcall’
[Sep 10 14:43:43] VERBOSE[9792] chan_sip.c: Scheduling destruction of SIP dialog '4b6731171080cd90535468292516a176@sip.broadvoice.com’ in 6400 ms (Method: INVITE)
[Sep 10 14:43:43] VERBOSE[9792] chan_sip.c: set_destination: Parsing sip:XXXXXXXXXX@206.15.156.221 for address/port to send to
[Sep 10 14:43:43] VERBOSE[9792] chan_sip.c: set_destination: set destination to 206.15.156.221, port 5060
[Sep 10 14:43:43] VERBOSE[9792] chan_sip.c: Reliably Transmitting (no NAT) to 206.15.156.221:5060:
BYE sip:XXXXXXXXXX@206.15.156.221 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK41284e15;rport
Max-Forwards: 70
From: “XXXXXXXXXX” sip:XXXXXXXXXX@sip.broadvoice.com;tag=as0ffa0150
To: sip:XXXXXXXXXX@sip.broadvoice.com;tag=stvw
Call-ID: 4b6731171080cd90535468292516a176@sip.broadvoice.com
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.11
Authorization: Digest username=“XXXXXXXXXX”, realm=“BroadWorks”, algorithm=MD5, uri="sip:XXXXXXXXXX@206.15.156.221", nonce=“BroadWorksXgdxk4d9rTv9d4woBW”, response=“0f53969f059b21a38c964d912446d6e5”, qop=auth, cnonce=“0c06f66d”, nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
[Sep 10 14:43:43] VERBOSE[9792] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/100-00000123’ in macro ‘dialout-trunk’
[Sep 10 14:43:43] VERBOSE[9792] pbx.c: == Spawn extension (from-internal, XXXXXXXXXX, 6) exited non-zero on ‘SIP/100-00000123’
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:206.15.156.221:5060 —>
SIP/2.0 200 OK
Call-ID: 4b6731171080cd90535468292516a176@sip.broadvoice.com
CSeq: 104 BYE
From: “XXXXXXXXXX” sip:XXXXXXXXXX@sip.broadvoice.com;tag=as0ffa0150
To: sip:XXXXXXXXXX@sip.broadvoice.com;tag=stvw
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK41284e15;rport=5060
Content-Length: 0
<------------->
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c: — (7 headers 0 lines) —
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c: Really destroying SIP dialog ‘c6ccedb92608044c@172.16.0.124’ Method: BYE
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c: Really destroying SIP dialog '4b6731171080cd90535468292516a176@sip.broadvoice.com’ Method: INVITE
[Sep 10 14:43:50] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:1024 —>
<------------->
[Sep 10 14:43:50] VERBOSE[2318] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:1024 —>
REGISTER sip:voip.XXXXXX.com SIP/2.0
Via: SIP/2.0/UDP 172.16.0.124;branch=z9hG4bK66c97d6e67fb9aa9
From: “Home” sip:100@voip.XXXXXX.com;tag=7563c2d1f198020e
To: sip:100@voip.XXXXXX.com
Contact: sip:100@172.16.0.124
Supported: replaces, timer
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:voip.XXXXXX.com”, nonce=“78905060”, response="4c8e8b1bf182f7581728b25ae4057ce9"
Call-ID: ed20b0003ec0b7a9@172.16.0.124
CSeq: 250 REGISTER
Expires: 3600
User-Agent: Grandstream HT287 1.1.0.45 DevId 000b822193d2
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
<------------->
[Sep 10 14:43:50] VERBOSE[2318] chan_sip.c: — (14 headers 0 lines) —
[Sep 10 14:43:50] VERBOSE[2318] chan_sip.c: Sending to 172.16.0.124 : 5060 (no NAT)
[Sep 10 14:43:50] VERBOSE[2318] chan_sip.c:
<— Transmitting (NAT) to XXX.XXX.XXX.XXX:1024 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.124;branch=z9hG4bK66c97d6e67fb9aa9;received=XXX.XXX.XXX.XXX
From: “Home” sip:100@voip.XXXXXX.com;tag=7563c2d1f198020e
To: sip:100@voip.XXXXXX.com
Call-ID: ed20b0003ec0b7a9@172.16.0.124
CSeq: 250 REGISTER
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
I think broadvoice sucks from seeing a ton of issues in the past, but that aside, what kind of a router are you using?
- Jake
www.voipcitadel.com
I’m using Linksys WRT54GL running Tomato.
Grandstream HT286 ATA seems to be the issue here.
<— SIP read from UDP:XXX.XXX.XXX.XXX:1024 —>
BYE sip:XXXXXXXXXX@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP 172.16.0.124;branch=z9hG4bK5c37586dad4e2fc3
From: “Home” sip:100@voip.XXXXXX.com;tag=779ee87e889e987e
To: sip:XXXXXXXXXX@voip.XXXXXX.com;tag=as5db45778
Supported: replaces, timer
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri="sip:XXXXXXXXXX@XXX.XXX.XXX.XXX", nonce=“230fbcd8”, response="28ad81a7ede50afb37b2436ba5c09a87"
Call-ID: c6ccedb92608044c@172.16.0.124
CSeq: 52466 BYE
User-Agent: Grandstream HT287 1.1.0.45 DevId 000b822193d2
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
<------------->
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c: — (12 headers 0 lines) —
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c: Sending to XXX.XXX.XXX.XXX : 1024 (NAT)
[Sep 10 14:43:43] VERBOSE[2318] chan_sip.c:
<— Transmitting (NAT) to XXX.XXX.XXX.XXX:1024 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.124;branch=z9hG4bK5c37586dad4e2fc3;received=XXX.XXX.XXX.XXX
From: “Home” sip:100@voip.XXXXXX.com;tag=779ee87e889e987e
To: sip:XXXXXXXXXX@voip.XXXXXX.com;tag=as5db45778
Call-ID: c6ccedb92608044c@172.16.0.124
CSeq: 52466 BYE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
i wouldn’t disagree that router should have no issues at all.
Why do you suspect its the ata?
-Jake
www.voipcitadel.com
Because I have other SIP devices (other than Grandstream ATA) that doesn’t hangup after 17:28.
Also, my Asterisk server is in the cloud (no NAT) and my SIP devices are in a LAN.
I use tomatoes myself without issue. That’s a pretty good deduction…I would say focus on that device, maybe start with a firmware update or similar if it is available?
-Jake
www.voipcitadel.com
I have another Asterisk 1.4 (wo/ FreePBX) server running and that doesn’t seems to have this issue when using HT286 ATA.
Asterisk 1.6 (w/ FreePBX 2.8.) seems to be the root cause. Anyone aware of such configuration that could trigger BYE message at 17Mins and 28 seconds (1048 seconds)?
Similar post on FreePBX forum:
freepbx.org/forum/freepbx/us … fter-17-28
Anyone?
The INVITE is missing from the trace.
The BYE comes from the remote device, and is indistinguishable from a user clear.
It is not immediately after another possibly relevant event.
Therefore, pending any indication that there was something strange about the invite, I would say the problem was with the remote device.
Changing the following on the ATA, seems to have fixed this issue:
UAC Specify Refresher: UAS
UAS Specify Refresher: UAS
Force INVITE: YES