In addition, I have created a possibly larger dump of the issue, as below. Can someone help me determine what the problem is? Is there more information that I can provide? I am running libpri 1.2.2, zaptel 1.2.4 and asterisk 1.2.5:
gdb dump:
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1211937872 (LWP 16798)]
0x00207138 in pri_disconnect_timeout (data="" at q931.c:2619
2619 if (pri->debug & PRI_DEBUG_Q931_STATE)
(gdb) bt
#0 0x00207138 in pri_disconnect_timeout (data="" at q931.c:2619
#1 0x002013db in __pri_schedule_run (pri=0x8a04010, tv=0xb7c33e3c) at prisched.c:98
#2 0x00201446 in pri_schedule_run (pri=0x8a04010) at prisched.c:110
#3 0x001d1282 in pri_dchannel (vpri=0x1e0c40) at chan_zap.c:8190
#4 0x004eb341 in start_thread () from /lib/tls/libpthread.so.0
#5 0x004406fe in clone () from /lib/tls/libc.so.6
(gdb) list
8190 e = pri_schedule_run(pri->dchans[which]);
8191 if (e)
8192 break;
8193 }
8194 } else if (res > -1) {
8195 for (which=0;which<NUM_DCHANS;which++) {
8196 if (!pri->dchans[which])
8197 break;
8198 if (fds[which].revents & POLLPRI) {
8199 /* Check for an event */
console dump (phone numbers have been removed)
Mar 6 08:58:19 VERBOSE[16799] logger.c: -- Channel 3/21, span 4 got hangup request
Mar 6 08:58:19 VERBOSE[16916] logger.c: -- Hungup 'Zap/3-1'
Mar 6 08:58:19 VERBOSE[16916] logger.c: == Spawn extension (macro-dialextNoCallid, s, 3) exited non-zero on 'Zap/93-1' in macro 'dialextNoCallid'
Mar 6 08:58:19 VERBOSE[16916] logger.c: == Spawn extension (macro-dialextNoCallid, s, 3) exited non-zero on 'Zap/93-1'
Mar 6 08:58:19 VERBOSE[16916] logger.c: -- Hungup 'Zap/93-1'
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Executing Macro("SIP/46583-82b2", "dialOutToronto|Zap/g1/9416xxxxxxx|ADP BROKER SVC|416xxxxxxx") in new stack
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Executing Answer("SIP/46583-82b2", "") in new stack
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Executing SetCallerID("SIP/46583-82b2", "ADP BROKER SVC <416xxxxxxx>") in new stack
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Executing Dial("SIP/46583-82b2", "Zap/g1/9416xxxxxxx") in new stack
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Requested transfer capability: 0x00 - SPEECH
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Called g1/9416xxxxxxx
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Zap/2-1 is proceeding passing it to SIP/46583-82b2
Mar 6 08:58:27 VERBOSE[17053] logger.c: -- Zap/2-1 is ringing
Mar 6 08:58:41 VERBOSE[17053] logger.c: -- Zap/2-1 answered SIP/46583-82b2
zaptel.conf
defaultzone=us
span=1,1,0,d4,b8zs
bchan=1-12
dchan=24
span=2,3,0,d4,b8zs
bchan=25-36
dchan=48
# span=3,0,0,esf,b8zs
# bchan=49-71
# dchan=72
span=4,2,0,esf,b8zs
bchan=73-95
dchan=96
zapata.conf
[trunkgroups]
trunkgroup => 1,24,48
trunkgroup => 2,96
spanmap => 1,1,0
spanmap => 2,1,1
spanmap => 4,2,3
[channels]
rxgain=8.0
txgain=-4.5
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
group = 1
context = trunk
usecallerid=yes
callerid = asreceived
switchtype = national
nsf = none
overlapdial = no
signalling = pri_net
channel => 1-12,25-36
rxgain=-5
group = 2
context = trunk
usecallerid=yes
callerid = asreceived
switchtype = national
overlapdial = no
signalling = pri_net
relaxdtmf = yes
channel => 73-95
------------------------------
Message: 4
Date: Sun, 5 Mar 2006 15:20:15 -0500
From: "McQuiggan, Mark xt46480" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Problem with libpri?
To: <asterisk-users@lists.digium.com>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="windows-1252"
While testing a problem with "spontaeously" and "occasionally" rebooting
asterisk, I came upon this problem:
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1210770512 (LWP 11346)]
0x002e3fe1 in pri_release_timeout (data="" at q931.c:2589
2589 q931.c: No such file or directory.
in q931.c
q931.c is in libpri, function pri_release_timeout, and line 2589 reads:
if (pri->debug & PRI_DEBUG_Q931_STATE)
pri_message(pri, "Timed out looking for release
complete\n");
PRI Debug was not on in the asterisk console.
Any ideas? My asterisk restarts about twice a day, and drops any current
calls in the process.
Regards,
Mark McQuiggan
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