[asterisk-users] asterisk core dumps on a Sipura forwarded to a
queue/moh
Vahan Yerkanian
vahan at arminco.com
Wed Jul 19 09:16:11 MST 2006
Greetings all,
I'm running Asterisk 1.2.9.1 installed from /usr/ports/net/asterisk on
FreeBSD 6.1-RELEASE.
I'm experiencing a guaranteed asterisk core dump with any Sipura device
set to forward all calls to an extension that is mapped to a queue:
-- Executing Macro("SIP/10040-4c43", "call|10027") in new stack
-- Executing Set("SIP/10040-4c43", "ext=10027") in new stack
-- Executing Dial("SIP/10040-4c43", "SIP/10027|20|o") in new stack
-- Called 10027
-- Got SIP response 302 "Moved Temporarily" back from 10.20.30.40
-- Now forwarding SIP/10040-4c43 to 'Local/111 at Main' (thanks to
SIP/10027-4f37)
sip*CLI>
Disconnected from Asterisk server
#
so the 10027 is the Sipura-3000 in this case, with configured "Cfwd All
Dest:" (forward all calls) to the extension 111, which is a queue or
109, which is a musiconhold call.
-rw------- 1 root wheel 11292672 Jul 19 21:15 asterisk.core
(gdb) bt
#0 0x2829d4ab in pthread_testcancel () from /usr/lib/libpthread.so.2
#1 0x28295e3c in pthread_mutexattr_init () from /usr/lib/libpthread.so.2
#2 0x2810a450 in ?? ()
(gdb) bt full
#0 0x2829d4ab in pthread_testcancel () from /usr/lib/libpthread.so.2
No symbol table info available.
#1 0x28295e3c in pthread_mutexattr_init () from /usr/lib/libpthread.so.2
No symbol table info available.
#2 0x2810a450 in ?? ()
No symbol table info available.
If I set the "Cfwd All Dest:" in the Sipura configuration interface to a
phone extension (f.e. 10011) everything works ok.
Any clue what's causing this?
--8<-- extensions.ael ---8<--
context default {
s => Goto(MainIVR|s|1);
};
context Main {
includes {
Gateways;
MainENUM;
};
s => Goto(MainIVR|s|1);
101 => Queue(InfoDesk);
111 => Queue(Support);
121 => Queue(Accounting);
131 => Queue(Admin);
141 => Queue(DomHosting);
};
--8<-- extensions.ael ---8<--
--8<-- queues.conf ---8<--
[Support]
timeout=60
context=Main
wrapuptime=15
announce-frequency=30
announce-holdtime=yes
monitor-format=wav49
monitor-join=yes
member => SIP/10061
member => SIP/10062
member => SIP/10063
--8<-- queues.conf ---8<--
Here is the sip debug:
-- Executing Macro("SIP/10040-681e", "call|10027") in new stack
-- Executing Set("SIP/10040-681e", "ext=10027") in new stack
-- Executing Dial("SIP/10040-681e", "SIP/10027|20|o") in new stack
-- SIP Seeding peer from astdb: '10027' at 10027 at 10.20.30.40:5060
for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.20.30.40:5060:
OPTIONS sip:10027 at 10.20.30.40:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK41db2c78;rport
From: "Unknown" <sip:Unknown at 10.20.30.1>;tag=as3340d5e5
To: <sip:10027 at 10.20.30.40:5060>
Contact: <sip:Unknown at 10.20.30.1>
Call-ID: 3ff6ab385b630a56433bc2f9087d8beb at 10.20.30.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 19 Jul 2006 16:02:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 3 lines
Reliably Transmitting (no NAT) to 10.20.30.40:5060:
NOTIFY sip:10027 at 10.20.30.40:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK7776a171;rport
From: "Unknown" <sip:Unknown at 10.20.30.1>;tag=as0d613efd
To: <sip:10027 at 10.20.30.40:5060>
Contact: <sip:Unknown at 10.20.30.1>
Call-ID: 18f6b3596c78a4dc648f9f261119ab5e at 10.20.30.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 94
Messages-Waiting: no
Message-Account: sip:asterisk at 10.20.30.1
Voice-Message: 0/1 (0/0)
---
Scheduling destruction of call
'18f6b3596c78a4dc648f9f261119ab5e at 10.20.30.1' in 15000 ms
We're at 10.20.30.1 port 10656
Video is at 10.20.30.1 port 15268
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to 10.20.30.40:5060:
INVITE sip:10027 at 10.20.30.40:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK723d1311;rport
From: "John Doe" <sip:10040 at 10.20.30.1>;tag=as5214182e
To: <sip:10027 at 10.20.30.40:5060>
Contact: <sip:10040 at 10.20.30.1>
Call-ID: 1773dd8737d4d6187633de4a474708e6 at 10.20.30.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 19 Jul 2006 16:02:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 9210 9210 IN IP4 10.20.30.1
s=session
c=IN IP4 10.20.30.1
t=0 0
m=audio 10656 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 10027
sip*CLI>
<-- SIP read from 10.20.30.40:5060:
SIP/2.0 200 OK
To: <sip:10027 at 10.20.30.40:5060>;tag=31180d12ce1539b5i0
From: "Unknown" <sip:Unknown at 10.20.30.1>;tag=as3340d5e5
Call-ID: 3ff6ab385b630a56433bc2f9087d8beb at 10.20.30.1
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK41db2c78
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
--- (10 headers 0 lines)---
Destroying call '3ff6ab385b630a56433bc2f9087d8beb at 10.20.30.1'
sip*CLI>
<-- SIP read from 10.20.30.40:5060:
SIP/2.0 200 OK
To: <sip:10027 at 10.20.30.40:5060>;tag=31180d12ce1539b5i0
From: "Unknown" <sip:Unknown at 10.20.30.1>;tag=as0d613efd
Call-ID: 18f6b3596c78a4dc648f9f261119ab5e at 10.20.30.1
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK7776a171
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '18f6b3596c78a4dc648f9f261119ab5e at 10.20.30.1'
sip*CLI>
<-- SIP read from 10.20.30.40:5060:
SIP/2.0 100 Trying
To: <sip:10027 at 10.20.30.40:5060>
From: "John Doe" <sip:10040 at 10.20.30.1>;tag=as5214182e
Call-ID: 1773dd8737d4d6187633de4a474708e6 at 10.20.30.1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK723d1311
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
--- (8 headers 0 lines)---
sip*CLI>
<-- SIP read from 10.20.30.40:5060:
SIP/2.0 302 Moved Temporarily
To: <sip:10027 at 10.20.30.40:5060>;tag=31180d12ce1539b5i0
From: "John Doe" <sip:10040 at 10.20.30.1>;tag=as5214182e
Call-ID: 1773dd8737d4d6187633de4a474708e6 at 10.20.30.1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK723d1311
Contact: <sip:109 at sip.Main.com>
Diversion: Anonymous <sip:10027 at sip.Main.com>;reason=unconditional
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
--- (10 headers 0 lines)---
-- Got SIP response 302 "Moved Temporarily" back from 10.20.30.40
Transmitting (no NAT) to 10.20.30.40:5060:
ACK sip:10027 at 10.20.30.40:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK723d1311;rport
From: "John Doe" <sip:10040 at 10.20.30.1>;tag=as5214182e
To: <sip:10027 at 10.20.30.40:5060>;tag=31180d12ce1539b5i0
Contact: <sip:10040 at 10.20.30.1>
Call-ID: 1773dd8737d4d6187633de4a474708e6 at 10.20.30.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
sip*CLI>
Disconnected from Asterisk server
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