hangup after 17:28 (1048 secs)

Get help with installing, upgrading and running Asterisk.

Moderators: muppetmaster, Moderator, Support

hangup after 17:28 (1048 secs)

Postby km » Fri Sep 03, 2010 5:34 pm

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?
km
Newsterisk
 
Posts: 8
Joined: Fri Jul 16, 2010 9:33 am

Re: hangup after 17:28 (1048 secs)

Postby david55 » Mon Sep 06, 2010 3:51 am

Not without the sip debug logs.
david55
Moves Like Spencer
 
Posts: 12570
Joined: Fri Sep 26, 2008 5:03 am

Re: hangup after 17:28 (1048 secs)

Postby voipcitadel.com » Mon Sep 06, 2010 6:31 pm

post a screenshot

-Jake
www.voipcitadel.com
voipcitadel.com
Oldsterisk
 
Posts: 82
Joined: Sun Sep 05, 2010 7:55 pm

Re: hangup after 17:28 (1048 secs)

Postby mjonescalpoly » Fri Sep 10, 2010 4:40 pm

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
mjonescalpoly
Newsterisk
 
Posts: 7
Joined: Fri Sep 10, 2010 4:27 pm

Re: hangup after 17:28 (1048 secs)

Postby voipcitadel.com » Sat Sep 11, 2010 12:48 am

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
voipcitadel.com
Oldsterisk
 
Posts: 82
Joined: Sun Sep 05, 2010 7:55 pm

Re: hangup after 17:28 (1048 secs)

Postby mjonescalpoly » Sat Sep 11, 2010 9:40 am

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
mjonescalpoly
Newsterisk
 
Posts: 7
Joined: Fri Sep 10, 2010 4:27 pm

Re: hangup after 17:28 (1048 secs)

Postby voipcitadel.com » Sat Sep 11, 2010 11:08 am

i wouldn't disagree that router should have no issues at all.

Why do you suspect its the ata?

-Jake
www.voipcitadel.com
voipcitadel.com
Oldsterisk
 
Posts: 82
Joined: Sun Sep 05, 2010 7:55 pm

Re: hangup after 17:28 (1048 secs)

Postby mjonescalpoly » Sat Sep 11, 2010 12:11 pm

Because I have other SIP devices (other than Grandstream ATA) that doesn't hangup after 17:28.
mjonescalpoly
Newsterisk
 
Posts: 7
Joined: Fri Sep 10, 2010 4:27 pm

Re: hangup after 17:28 (1048 secs)

Postby mjonescalpoly » Sat Sep 11, 2010 12:16 pm

Also, my Asterisk server is in the cloud (no NAT) and my SIP devices are in a LAN.
mjonescalpoly
Newsterisk
 
Posts: 7
Joined: Fri Sep 10, 2010 4:27 pm

Re: hangup after 17:28 (1048 secs)

Postby voipcitadel.com » Sat Sep 11, 2010 2:09 pm

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
voipcitadel.com
Oldsterisk
 
Posts: 82
Joined: Sun Sep 05, 2010 7:55 pm

Re: hangup after 17:28 (1048 secs)

Postby mjonescalpoly » Sat Sep 11, 2010 5:40 pm

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:
http://www.freepbx.org/forum/freepbx/us ... fter-17-28
mjonescalpoly
Newsterisk
 
Posts: 7
Joined: Fri Sep 10, 2010 4:27 pm

Re: hangup after 17:28 (1048 secs)

Postby mjonescalpoly » Mon Sep 13, 2010 8:33 am

Anyone?
mjonescalpoly
Newsterisk
 
Posts: 7
Joined: Fri Sep 10, 2010 4:27 pm

Re: hangup after 17:28 (1048 secs)

Postby david55 » Mon Sep 13, 2010 8:53 am

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.
david55
Moves Like Spencer
 
Posts: 12570
Joined: Fri Sep 26, 2008 5:03 am

Re: hangup after 17:28 (1048 secs)

Postby mjonescalpoly » Thu Sep 16, 2010 10:38 am

Changing the following on the ATA, seems to have fixed this issue:

UAC Specify Refresher: UAS
UAS Specify Refresher: UAS
Force INVITE: YES
mjonescalpoly
Newsterisk
 
Posts: 7
Joined: Fri Sep 10, 2010 4:27 pm


Return to Asterisk Support

Who is online

Users browsing this forum: Exabot [Bot] and 1 guest