Public bug reported:
An example of the failure: http://logs.openstack.org/91/189391/6/check
/check-neutron-dsvm-functional/0ba6e51/console.html
A logstash query:
I believe this has been fixed by:
https://review.openstack.org/#/c/165117/
** Changed in: neutron
Status: In Progress = Fix Released
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Right now I'm leaning toward parent always does nothing + PluginWorker.
Everything is forked, no special case for workers==0, and explicit
designation of the only one case. Of course, it's still early in the day
and I haven't had any coffee.
I have updated the patch
There are two classes of behavior that need to be handled:
1) There are things that can only be done after forking like setting up
connections or spawning threads.
2) Some things should only be done once regardless of number of forks, like
syncing.
Even when you just want something to happen
/rpc workers != 0.
** Affects: neutron
Importance: Undecided
Assignee: Terry Wilson (otherwiseguy)
Status: In Progress
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- Original Message -
On Wed, May 27, 2015 at 12:11:55AM -0400, Russell Bryant wrote:
On 05/26/2015 10:14 PM, Ben Pfaff wrote:
On Tue, May 26, 2015 at 10:31:57AM -0400, Russell Bryant wrote:
On 05/21/2015 10:23 PM, Andy Hill wrote:
As a consequence, this requires dropping
This patch fixes just the Python 3 problems found by running
python3 setup.py install
There are still many other issues to be fixed, but this is a start.
Signed-off-by: Terry Wilson twil...@redhat.com
---
python/compat/argparse.py | 2 +-
python/ovs/daemon.py | 31
Importance: Undecided
Assignee: Terry Wilson (otherwiseguy)
Status: In Progress
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Title:
The native OVSDB Connection class
on an update to a Row object to make it easy for
users of notify() to see what changed, though this usage of Row
is quite different than Idl's typical use.
Signed-off-by: Terry Wilson twil...@redhat.com
---
python/ovs/db/idl.py | 42 +-
tests/ovsdb-idl.at
on an update to a Row object to make it easy for
users of notify() to see what changed, though this usage of Row
is quite different than Idl's typical use.
Signed-off-by: Terry Wilson twil...@redhat.com
---
python/ovs/db/idl.py | 42 +-
tests/ovsdb-idl.at
on an update to a Row object to make it easy for
users of notify() to see what changed, though this usage of Row
is quite different than Idl's typical use.
Signed-off-by: Terry Wilson twil...@redhat.com
---
python/ovs/db/idl.py | 42 +-
1 file changed, 41
This adds very basic support for setuptools so that the OVS Python
lib can be added to PyPI.
This currently uses the Open vSwitch version number and the
generated dirs.py, though there is no real reason to tie the
Python libraries releases or version numbers to the main project's.
---
Reposting
Signed-off-by: Terry Wilson twil...@redhat.com
/me sighs
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- Original Message -
On 04/10/2015 03:57 PM, Terry Wilson wrote:
This adds very basic support for setuptools so that the OVS Python
lib can be added to PyPI.
Most Python libraries are on PyPI, so it makes sense to put this one
there, too. Another specific reason this would
Public bug reported:
A db_get call for a column that contains UUIDs will return one or more
Row objects instead of one or more UUIDs like the ovs-vsctl
implementation.
** Affects: neutron
Importance: Undecided
Assignee: Terry Wilson (otherwiseguy)
Status: In Progress
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Public bug reported:
ovs-vsctl supports looking up records in some tables by non-indexed
columns.
For example,
attr = [('connection-mode', connection_mode)]
self.ovsdb.db_set('Controller', self.br_name, *attr).execute(
check_error=True)
with the vsctl backend will run:
ovs-vsctl
Public bug reported:
It is possible for the arping process to hang if the interface is
removed while it is running. An example script to generate a hung arping
process is attached.
** Affects: neutron
Importance: Undecided
Assignee: Terry Wilson (otherwiseguy)
Status
Sorry, I dropped the ball here. This is now released.
Unfortunately, the new novaclient release ended up completely breaking the
neutron gate. The v1_1 deprecation broke our (voting) pylint test:
https://jenkins04.openstack.org/job/gate-neutron-pylint/1383/console
2015-02-19 18:37:06.932 |
- Original Message -
On Feb 19, 2015, at 11:52, Terry Wilson twil...@redhat.com wrote:
Unfortunately, the new novaclient release ended up completely breaking the
neutron gate. The v1_1 deprecation broke our (voting) pylint test:
https://jenkins04.openstack.org/job/gate-neutron
** Also affects: neutron
Importance: Undecided
Status: New
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Title:
processutils checks whether stdlib is monkey patched
Public bug reported:
The ovs-vsctl 'list' command can take a list of records as an argument,
so there is no need to manually loop through all records discarding the
ones with names that don't match the bridge's port name list.
** Affects: neutron
Importance: Undecided
Assignee: Terry
Assignee: Terry Wilson (otherwiseguy)
Status: In Progress
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Title:
The L3 agent tries to catch an exception from processutils when
dhcp process on the system to restart. On systems with lots of networks
this can easily take longer than the default resync timeout leading to a
system that becomes unresponsive because of the load continually
restarting causes.
** Affects: neutron
Importance: Undecided
Assignee: Terry
This allows things like initiating a wait request on an interface
ofport being set.
When the optional field is empty and operation is != or excludes
then the result is true; otherwise it is false. If the field is
set then the field is compared normally for its type.
Signed-off-by: Terry Wilson
an exception on retry expiration also seems like a good idea.
** Affects: neutron
Importance: Undecided
Assignee: Terry Wilson (otherwiseguy)
Status: In Progress
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Public bug reported:
check-tempest-dsvm-neutron-2 failing with:
2014-06-19 22:16:03.198 | 2014-06-19 22:07:37,376 Failed to establish
authenticated ssh connection to cirros@172.24.4.71 ([Errno 111] Connection
refused). Number attempts: 4. Retry after 5 seconds.
2014-06-19
- Original Message -
What's the progress by Terry Wilson?
If not much, I'm willing to file blueprint/spec and drive it.
thanks,
I've been working on some proof-of-concept code to help flesh out ideas for
writing the spec. I'd talked to Maru and he mentioned that he didn't think
ipsec -j ACCEPT ' True False
** Affects: neutron
Importance: Undecided
Assignee: Terry Wilson (otherwiseguy)
Status: In Progress
** Tags: icehouse-backport-potential vpnaas
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: Undecided
Assignee: Terry Wilson (otherwiseguy)
Status: In Progress
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Title:
Nova notification introduces a hard
A question about the fix from https://review.openstack.org/#/c/82931
Also, how does this work for RHEL-based distros where they tend to backport
new kernel features? For instance vxlan support was added in the kernel for
RHEL6.5 which is 2.6.32-based... That changeset looks like it breaks
Public bug reported:
fwaas_driver.ini is missing from setup.cfg
** Affects: neutron
Importance: Undecided
Assignee: Terry Wilson (otherwiseguy)
Status: In Progress
** Tags: havana-backport-potential
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Public bug reported:
Both vpnaas.filters and debug.filters are missing from setup.cfg,
breaking rootwrap for the appropriate commands.
** Affects: neutron
Importance: Undecided
Assignee: Terry Wilson (otherwiseguy)
Status: In Progress
** Tags: havana-backport-potential low
I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a
flood of:
WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL
logic
error or missing database
AFAIK, I'm not doing any database puts (or gets). There were no such
warnings in 1.8.8.0.
What do I need to do
I have not looked at the log files, but often times DSL routers may use PPPoE
which has a little bit of overhead so you need to set the MTU below the default
of 1500. Some info about the issue can be found here:
http://www.ezlan.net/PPPOE.html and
I managed to get it working eventually. I think that it may have been
a problem with neon , as I downgraded to .25 from .29, removed all
modules and make distclean, make install
It started working at this point !
Good to hear.
What would be really great would be
1) manager events for
I am trying to get googleapps calendar integrated with my system.
However, following all the instructions that I can find it still
fails. this is my config file:
[myGoogleCal]
type=caldav
url=https://www.google.com/calendar/dav/myemail/events/
user=myemail
secret=mypassword
refresh=15
So its quite confusing. Why it dont fill the ipaddr field?? From which
SIP message get and cut out the IP address?
Realtime updates the ipaddr field when a phone registers as that peer.
--
_
-- Bandwidth and Colocation
Since when can someone submit a patch for chan_skype?? Did i miss an
announcement that it has been opensourced? I'm under the impression
that digium is the only party who *can* extend chan_skype...
Paul was a little confused and thought something would have to be added to
Asterisk. But, with
I have canreinvite and directmedia to 'no' - and there is no NAT
between the phones and asterisk...
Hmm. In that case, I'm not sure. You could take a look at the output of rtp
set debug on when the call is going on to see what is going on with the audio.
--
Hello,
I have a strange audio delay behaviour when placing a call between two
SIP devices using the same codec.
In my example, I have two devices forced to use GSM codec.
When placing a call, the first ~9sec have no audio, then the audio
starts trasmitting.
If I force one phone to use GSM
What does this mean ? What can I do further ?
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
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New to Asterisk? Join us for a live
our own agi application. Can we use the native database connectors for
ARA. I currently have
everything working with unixodbc + myodbc however, looking to use the
native DB connector
if possible.
Traditionally, the mysql realtime backend was buggy and crash-prone. The odbc
backend is has a
I do not know anything about 10.0 but 1.6.2 problem most likely can be
fixed by a simple patch which is not being committed for unknown
reason
since late August 2011.
https://issues.asterisk.org/jira/browse/ASTERISK-18301?focusedCommentId=183734#comment-183734
1.6.2 is in security fix only
- Original Message -
From: Sam Muro resea...@businesstz.com
To: asterisk-users@lists.digium.com
Sent: Friday, October 14, 2011 2:02:01 AM
Subject: [asterisk-users] Asterisk Security: Allow only one phone per sip
registration
Hi there
Consider this. You have three SIP extension
Is there a way one can bind sip account to specific mac-address
(assume on
the same subnet). In this way, even if you know the username/secret,
you
will still have to use the same physical phone, unless you play with
mac-address.
No. And mac addresses are easily spoofed so it would not
Thanks. Let me see how best i can complicate them per phone. Ooops,
1000
sip phones
If it were me, I would look into Asterisk Realtime for handling the SIP phones.
I would then write a script to generate the configs for the phones into the SIP
realtime database with random passwords. Match
It was not intentional, probably a side-effect of the switch to SQlite 3
from BDB. Unfortunately, that command was not documented to produce the
database results ordered in any particular order, so this change isn't a
bug, just a side-effect.
Thanks. The only time it really matters to me
This bug is fixed by the Debian dragonegg-2.8-3 package as described
here: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=618768
This bug makes the package completely unusable for compiling.
** Bug watch added: Debian Bug tracker #618768
debug logs below:
Asterisk 1.8.4: http://pastebin.com/DFnKgSse
Asterisk trunk r319661: http://pastebin.com/B19tdbxJ
These show that a proper bridging tech module cannot be found to run
ConfBridge. The debug message showing that a capability for ulaw couldn't be
found was a buggy debug
2455 South Road
Poughkeepsie, NY 12601
(845) 435-8666, T/L 295-8666
rayhi...@us.ibm.com
The IBM z/VM Operating System IBMVM@LISTSERV.UARK.EDU wrote on
04/12/2011 05:55:53 PM:
From: Terry Wilson terry.wil...@potashcorp.com
To: IBMVM@LISTSERV.UARK.EDU
Date: 04/12/2011 05:56 PM
Subject: Re
Mark,
I should have mentioned the reply from the command
zlinux1:~ # lszfcp -D
0.0.5400/0x50050763070342f3/0x4010400b 0:0:1:1074479120
is from the 8300 storage and works fine. Its all the other WWPN's that
are having the problem
From: Mark Post mp...@novell.com
To:
Subject:Re: Cannot see SVC disk zLinux
Sent by:The IBM z/VM Operating System IBMVM@LISTSERV.UARK.EDU
The IBM z/VM Operating System IBMVM@LISTSERV.UARK.EDU wrote on
04/13/2011 10:15:44 AM:
From: Terry Wilson terry.wil...@potashcorp.com
To: IBMVM@LISTSERV.UARK.EDU
Date: 04/13
-0
Hello,
I am having a problem connecting ZFCP through my SVC. When I issue the
lsluns command it comes back with Unable to send the REPORT_LUNS command
to LUN. Any ideas?
Attached are commands
zlinux1:~ # lszfcp -P(
0.0.5400/0x500507680140a23a rport-0:0-0
0.0.5400/0x50050763070342f3
handling,
but who knows... maybe there is a bug.
Regards,
Ray Higgs
System z FCP Development
Bld. 706, B42
2455 South Road
Poughkeepsie, NY 12601
(845) 435-8666, T/L 295-8666
rayhi...@us.ibm.com
The IBM z/VM Operating System IBMVM@LISTSERV.UARK.EDU wrote on
04/12/2011 04:14:43 PM:
From: Terry
On 03/03/2011 02:22 PM, Mitch Johnson wrote:
Thanks so much for pointing this out. I was curious why the commands in the
documentation differed to the commands I was using.
That problem is fixed, but now I have a new issue. I can call with no issues,
however, as soon as I answer one of the
On Feb 28, 2011, at 7:19 PM, mitch Johnson wrote:
I'm in the process of testing a TLS/SRTP install. My experience is improving
with each new challenge, but this one is a great test of my 2 month
experience with Asterisk.
[myphones]
;exten = 6001,1,Dial(SIP/6001)
;exten =
On Mar 1, 2011, at 4:19 AM, Ishfaq Malik wrote:
Seeing that 1.8.3 had been released I updated our main test server to
that from 1.8.2.2 using the digium yum repo.
All audio had been working fine on this server before the update but
after the update I experienced the same as I did with
Hi,
I'm wondering if this is normal asterisk behaviour:
asterisk*CLI sip show channels
Peer User/ANR Call ID Format Hold
Last MessageExpiry Peer
10.12.0.2(None) 3c2f7ff2975e-wp 0x0 (nothing)No Rx:
PUBLISH
On Feb 28, 2011, at 5:02 PM, isr...@gmail.com wrote:
As far I know asterisk doesn't handle the publish sip dialog so it just keeps
it hanging around in 1.8.X (in previous versions it didn't)
Asterisk 1.8 does handle PUBLISH dialogs, which is why they stay around.
--
Apologies in advance if this has come up a thousand times before but is there
any way to stop this error in 1.8 ?
[ Feb 26 15:09:09] ERROR[6678] chan_sip.c: No SRTP module loaded, can't setup
SRTP session.
Sounds like your hone is sending an SRTP offer, but you don't have res_srtp
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received
Can you please send me a how to please or a simple lines?
Regards
Please see the README file that came with skypeforaterisk. Search for
SkypeChatMessage.
As far as AMI tutorial, please see Asterisk: The Definitive Guide chapter 20
(and consider ordering a copy).
On Feb 23, 2011, at 4:39 AM, Ishfaq Malik wrote:
Hi
We're still testing out asterisk 1.8 (using 1.8.2.2 from rpm package)
before putting it into production and I'm observing an odd issue when
using the AMI
Every request I send to the AMI just results in a FullyBooted response
rather
Hi,
I have this same behaviour on version 1.8.2.3 build from source. We are using
AMI to originate call from our CRM software, but we ignore that message.
The patch for the bug at https://issues.asterisk.org/view.php?id=18168 has been
committed (thanks FeyFre!). The FullyBooted event will
On Feb 23, 2011, at 7:11 PM, Jose P. Espinal wrote:
On 02/23/2011 08:56 PM, Leif Madsen wrote:
Actually I was wrong!
See here. It is being resolved.
https://reviewboard.asterisk.org/r/1107/
Leif.
Thanks for the feedback, Leif!
I will follow that incident closely, as I was
and must be closed.
Hope you can help,
Regards,
Terry Wilson
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we have a similar problem. When we try to make two skype-calls at a time,
only one of them has working audio. For this to happen, both calls must be
ringing at the same time. Does anyone know how to fix this?
I have fixed this issue and it will be in the 1.0.7 release which is currently
in
You need to compile sipp with pcap support. Here is an example
scenario: http://sipp.sourceforge.net/doc3.0/reference.html#UAC+with+media
On Sep 22, 2009, at 5:13 AM, DHAVAL INDRODIYA wrote:
Hello
I would like to play file with sipp command.
I want to take value of RTPAUDIOQOS for every
Also, you can pass -sn uac_pcap to use the default pcap_play scenario
which will play 8 seconds of audio and a dtmf digit. You can also run
sipp -sd uac_pcap uac_pcap.xml to save the scenario and edit it.
You can then play it back using sipp -sf uac_pcap.xml ...
On Aug 20, 2009, at 3:41 AM, Remco Barendse wrote:
I never used Skype myself but i installed it to try and i noticed
that i
got added by lots of strange skype users (spam bots?), i guess some of
those were trying some funny stuff on my skype for asterisk account. I
want to use Skype for
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity to actually fix any bugs
that people find. :-)
I installed the 1.0 release of Skype for Asterisk and last night on my
production box running
That does sound a bit pricey, although it it's as stable as the latest
beta, I wont be buying it at all.
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity to actually fix any bugs
that people find.
I just want to also remind people that Skype for SIP is also to be
released shortly. When I last talked to Skype they said it would be
out in late July. So I assume if you wait another few more weeks
the entire issue will be moot. No $60/channel fee, just the free
SIP platform for
hello,
In last november was reviewed and committed code that enabled
Asterisk to read from or write to iCal, CalDAV and Exchange calendars.
1. Has anyone successfully used this feature with an Exchange
server ? Is it easy to create an Exchange login allowing Asterisk to
query and
I have problems with it...
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1153 skype_cp_handler:
Found license 'XX' providing 1 concurrent calls
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1000 display_host: Skype
For Asterisk Host-ID: X
[Jul 30 14:34:17] NOTICE[30529]:
photoshop
'export for the web'.
Is there a way to mimic that with imagemagick?
Thanks!
S.
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Terry Wilson | te...@terryfic.com | http
Of course you should be using Lua.
I really have to try that sometime
tAt the same time, traditional .conf dialplans
are not going away anytime soon, and you do not lose any functionality
vs AEL and Lua.
My reason for sticking with .conf files so far? dialplan show -- it
is easier
This has been on my ToDo list far too long.
I have a small call-center setup, with basic
time of day/day of week validation before putting
callers in the queues.
Just thought I'd post here that this is one of the reasons I developed
Calendaring for Asterisk. Basically, you can use
I have seen bug http://bugs.digium.com/view.php?id=13525 - i think
it is
releated to it
I talked to file (who closed the above bug) and he thinks that it is
probably a different issue with similar symptoms. If you could go
ahead and post a bug on bugs.digium.com for it, that'll give us
On Dec 15, 2008, at 7:05 AM, Mike wrote:
Just so I'm clear: there is no way to do what I want short of
playing with the underlying code, correct?
Yes. I'm working on an issue right now related to parking and noticed
that Asterisk completely lies with the verbose statement saying that
Thanks for the answer Terry, it's kind of what I expected. I may
have to look into using Attended transfers in Asterisk, but I think
my users really prefer having the TRNSFR soft key instead of
remembering a feature code.
Just for fun, I added the functionality that you wanted in
Is there any way to provide the user receiving an attended transfer
with a tone or other audible indication that the transfer is
completed (i.e. Party A calls Party B, Party B announces the call
while transferring to Party C, Party C hears tone when Party B
completes the transfer so
Yehavi Bourvine wrote:
OK, but I still did not get a reply to my original question: Why
using
SIP registrar in front of Asterisk and not simply use bare Astersik?
can't it handle the load? (remember - in my case it doesn't handle
the
RTP, only signalling). Can't it handle so much
I've looked at doing various things to chan_sip to improve signaling
performance (hash tables for call lookups, etc.) I gave up when I
realized that the overhead of handling the RTP was so far above the
overhead of processing SIP signaling that it didn't really matter
much. The only reason
2) Also wondering what people do when parsing asterisk verbose
output in the log. Specifically, following a certain call.
Asterisk's verbose output logs in sequence of action, which is good,
but if you have 40-50 workstations going at once, tracking the
progress of one call you are
hi
for any context ,you must to open /etc/asterisk/extensions.conf and
insert this line : exten =Realtime/[EMAIL PROTECTED]
and (reload) or (restart now) your asterisk
You don't have to restart asterisk, just a 'dialplan reload' will
suffice. So really there is no impact to a running
when i wnat to working with realtime and mysql
for any context i have to insert (switch = Realtiem/
[EMAIL PROTECTED]) statment into extensions.conf
See http://bugs.digium.com/view.php?id=6019
It looks like this will be available in Asterisk 1.6.1. You can try
out the beta if you would
I'm trying to find a way to generate 484 Address Incomplete SIP
response based on the
length of the extension called.
See the allowoverlap option in sip.conf. It should cause a 484 to be
sent if the address is potentially matchable, but not yet matching.
How can we add new contexts in asterisk realtime module? All I could
figure out after googling is that a new context HAS to be declared
in extensions.conf with 'switch = Realtime/@databasetable' under
the context name declaration. This works fine as long as we are
adding extensions
The setup is as follows: SIP phone registers via international link
to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2
via Zaptel Channels need to be hairpinned from Box 1 to 2. How is
sip.conf configured on Box 1 and 2 so that we don't get an error:
Failed to
I noticed the message in the archives
http://lists.freedesktop.org/archives/avahi/2006-September/000864.html
about an intention to dual license the avahi client libraries under
LGPL/MIT-X11. Was this ever done?
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Digium has released version 3.0.3 of its product registration
utility. This is the first version of the registration utility that
is compiled against the uClibc C library. A benefit of this
transition is that the register binary should run more consistently
and reliably across a wider
Beyond that, looks like they rolled their own PBX and email. While
they mention open source, it looks like its only for libraries, but
not the total package (save the email client, as they do mention
Thunderbird):
http://www.unison.com/opensource/
It looks like they are using SIP
Maybe we can start creating a collection of SIPP tests to run various
scenarious.
I would like to test how registrations and subscriptions affect the
stack too.
I have written a tool in perl that allows you to take a SIP pcap
capture (live or saved dump) and generate a sipp scenario from
I guess the commit of TOUPPER and TOLOWER funcs was a mistake since
they are not related
to the commit at all.
They are in fact related. They are required for proper processing of
the Polycom phone profile in phoneprov.conf. The information is
documented in the TeX documentation. I know
When I run the utility register of asking me for the license
number. I
get the following.
[EMAIL PROTECTED] modules]# /root/register
Digium Product Registration - Version 3.0.0
Copyright (C) 2004-2007, Digium, Inc.
Use the '-l' option to see license information for software
What is the reason for such response?
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP
192.168.129.74
:5160;branch=z9hG4bK17c3.17db29e7.0;received=192.168.129.74
Via: SIP/2.0/UDP 192.168.129.74 ;branch=z9hG4bK17c3.23083974.0
Via: SIP/2.0/UDP
I'm working on a 500 seats Asterisk project.
I'm wondering whether or not I should consider using Asterisk
Realtime and a database to manage phones registrations.
As far as stability goes, I've had no problem with realtime. In fact
I've run a nationwide VoIP provider with asterisk using
On Dec 21, 2007, at 1:29 PM, Brian Capouch wrote:
Terry Wilson wrote:
config files, etc. Make your SIP usernames meaningless and use
func_odbc to look up what extension is tied to which device.
I wouldn't say to make the names meaningless, though; there are
different ways to use those
, and everything
seems to compile correctly. Thanks!
Terry Wilson
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