similar to: No sound when the call is diverted

Any suggestion.... On Dec 6, 2017 11:51, "ABHISHEK PALIWAL" <abhishpaliwal at gmail.com> wrote: > Hi Team, > > We are getting the crash in glusterd after start of it. When I tried to > debug in brick logs we are getting below errors: > > [2017-12-01 14:10:14.684122] E [MSGID: 100018] > [glusterfsd.c:1960:glusterfs_pidfile_update] 0-glusterfsd: pidfile >

If I have inbound SIP calls arriving from a provider's gateway to an asterisk server on my LAN, which then routes the call back out via the provider's gateway to a PSTN number, once the call is answered do all the voice packets pass through my asterisk PBX, or is SIP intelligent enough to patch the two PSTN ends of the call direct to each other going only via two ports on the

Without the glusterd log file and the core file or the backtrace I can't comment anything. On Wed, Dec 6, 2017 at 3:09 PM, ABHISHEK PALIWAL <abhishpaliwal at gmail.com> wrote: > Any suggestion.... > > On Dec 6, 2017 11:51, "ABHISHEK PALIWAL" <abhishpaliwal at gmail.com> wrote: > >> Hi Team, >> >> We are getting the crash in glusterd after

Hi Atin, Please find the backtrace and logs files attached here. Also, below are the BT from core. (gdb) bt #0 0x00003fff8834b898 in __GI_raise (sig=<optimized out>) at ../sysdeps/unix/sysv/linux/raise.c:55 #1 0x00003fff88350fd0 in __GI_abort () at abort.c:89 [**ALERT: The abort() might not be exactly invoked from the following function line. If the trail function

I hope these logs were sufficient... please let me know if you require more logs. On Wed, Dec 6, 2017 at 3:26 PM, ABHISHEK PALIWAL <abhishpaliwal at gmail.com> wrote: > Hi Atin, > > Please find the backtrace and logs files attached here. > > Also, below are the BT from core. > > (gdb) bt > > #0 0x00003fff8834b898 in __GI_raise (sig=<optimized out>) at >

Hi, sorry for cross-mailing. Reply-to: set to freebsd-net. I have seen some discussion on freebsd-security etc. about some parts of the subject. I have seen older messages in archives. Regularly the same questions seem to come up. I have not found an all-including description of the answer to s.th. like: "Can anybody tell me the order packets get processed in kernel related to IPSec,

Hi All, I am looking for a way to allow users to dial *21*, followed by a number and the pound key. Asterisk must then divert all incoming calls to the user's extension to the number given. I got it right halfway using a tip from http://ns1.jnetdns.de/jn/relaunch/asterisk/page6.html. Thanks to whomever is the contributor of the above tip. The problem is however, that the number to which

I have a TDM400 running telco lines on ZAP2-4 My after hours config is supposed to receive the incoming call then divert it to my home phone by calling out one of the other zap channels available. console output as such... Starting simple switch on 'Zap/3-1' -- Executing Dial("Zap/3-1", "ZAP/G1/0823274210") in new stack Aug 20 19:06:44 NOTICE[646]:

Dear List, I have a series of nested loops with the structure shown below, and I am struggling to figure out how to divert output to folders created with dir.create() within the loops. What I need is for the output to end up as topNameK/subNameL/objNameM.pdf; what I get instead is a series of directories topNameK/, directories subNameL/, and files objNameM.pdf, all in the working directory. Any

> On a related note, how do you get a Cisco 7940 to dial numbers with a > hash in them, instead of just using the hash as a dial key. For > example, I have *#21# to check diverts, but the phone will just dial > "*" as soon as you type the # after it. <DIALTEMPLATE> <TEMPLATE MATCH="#..." Timeout="5" User="Phone" /> <TEMPLATE

I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first number's call to Trunk if second number is unregistered. Is it possible? ?f yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567---------------------------- registered to Asterisk

I have a issue trying to understand why Asterisk-PBX, when a SNOM (320 or 360) successfully redirects/diverts a call when it is a local extension, but fails when you enter external number. Both the local extension dial and external extension dial are within the same context [from-sip] and both phones are capable of making external calls. I have looked at the standard sites, but not

--- src/chroot.ml | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/src/chroot.ml b/src/chroot.ml index b5c1e53..9e522d9 100644 --- a/src/chroot.ml +++ b/src/chroot.ml @@ -26,7 +26,9 @@ let build_chroot debug files outputdir = List.iter ( fun file -> try - let path = file.ft_source_path in + let path = if file_exists file.ft_source_path +

--- configure.ac | 1 + src/config.ml.in | 1 + src/dpkg.ml | 1 + 3 files changed, 3 insertions(+) diff --git a/configure.ac b/configure.ac index 2141540..99ea913 100644 --- a/configure.ac +++ b/configure.ac @@ -92,6 +92,7 @@ AC_PATH_PROG(APT_GET,[apt-get],[no]) AC_PATH_PROG(DPKG,[dpkg],[no]) AC_PATH_PROG(DPKG_DEB,[dpkg-deb],[no]) AC_PATH_PROG(DPKG_QUERY,[dpkg-query],[no])

While trying to run libguestfs tests after building with "--enable-appliance --with-supermin-extra-options=--use-installed", I ran into a peculiar error message in the c-api test: ,---- | libguestfs: error: strings: /abssymlink: strings: error while loading | shared libraries: libbfd-2.24-multiarch.so: cannot open shared object | file: No such file or directory `---- The problem here

I've been using ipfw for a while to create a router with NAT and packet filtering, but have never combined it with stateful filtering, instead using things like "established" to accept incoming TCP packets which are part of a conversation initiated from the "inside". I'd like to move to using keep-state/check-state to get tighter filtering and also to allow outgoing