On Wed, Nov 10, 2021 at 09:08:52AM +, Kingsley Tart wrote:
> my last few emails to this list haven't appeared so I'm just testing
1. Check the archive:
http://lists.digium.com/pipermail/asterisk-users/2021-November/thread.html
2. Check your list settings (e.g: Receive your own posts to the
my last few emails to this list haven't appeared so I'm just testing
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Testing again.
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Testing the new lists.digium.com server. Apologies for the email noise.
Digium IT
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com>
Ämne: [asterisk-users] Test
Testing, 1, 2, 3.
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Testing, 1, 2, 3.
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Matthew Fredrickson
Digium, Inc. | Engineering Manager
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This is a test.
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Just a test, my apologies.
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Hi,
Just checking if my emails reach the list.
Thanks,
Amanda
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Fail.
On 10/28/2015 04:42 PM, ama...@sevana.fi wrote:
Hi,
Just checking if my emails reach the list.
Thanks,
Amanda
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Hi List,
I'm trying to get working SIPp with media but something is wrong (it's
working well without media), please help:
This is the command I send at SIPp server:
./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err
This is the result I see:
Last Error: Aborting call
Hello Everyone,
I wonder if someone could share a manual about using SIPp for Asterisk's
testing.
I'll be gratefull
Regards,
Elder Arohuanca
Lima - Peru
On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe zac.wo...@gmail.com wrote:
Sipp looks pretty good! I don't know how I missed this one. This
dan, elder,
I have played with scripts to generate calls and track their
completion, email me off-list if you have questions.
daveC
Daniel - Asterisk wrote:
Hello Everyone,
I wonder if someone could share a manual about using SIPp for
Asterisk's testing.
I'll be gratefull
Regards,
Hi,
We are searching for a pool of test numbers to call from Asterisk, record voice
and test it with our non-intrusive voice quality testing software (NIQA). The
problem is that we could find some test numbers, but our customer would like to
have a global pool of test numbers, so that we can
Looking to see if it shows up - thanks
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hi:
only test
Best wishes!
Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7,
elastix, trixbox support.
website:www.cnasterisk.com, www.voip88.com
voip88
_
Test successful
On 2010-03-21 9:12 AM, card support asterisk asteriskc...@hotmail.com
wrote:
hi:
only test
Best wishes!
Asterisk Support group(sangoma, digium...), providing asterisk conf, pri,
ss7, elastix, trixbox support.
website:www.cnasterisk.com, www.voip88.com
voip88
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fail.
On Tue, 9 Feb 2010, aster...@opensourcesolution.in wrote:
test
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At 5:59 PM on 19 Jan 2010, __ wrote:
Test case:
We have e1 trunk and multi-channel sip line. Clients waiting in the
queue, which can handle 30 clients. They listen mellody and their
position, while waiting. The system can handle only 5 clients at the
moment. As soon
On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
cwall...@lodgingcompany.com wrote:
At 5:59 PM on 19 Jan 2010, __ wrote:
Test case:
We have e1 trunk and multi-channel sip line. Clients waiting in the
queue, which can handle 30 clients. They listen mellody and their
At 3:09 AM on 21 Jan 2010, __ wrote:
On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
cwall...@lodgingcompany.com wrote:
At 5:59 PM on 19 Jan 2010, __ wrote:
Test case:
We have e1 trunk and multi-channel sip line. Clients waiting in
Hello, list.
First of all i want to say sorry for my english.
Long story short, on my future work i'll deal with asterisk and now i
have a test case. But i'm very young to asterisk and don't have a lot
of time so any help is appreciated.
Test case:
We have e1 trunk and multi-channel sip line.
A new patch has been made for an extra Manager Event when a call-pickup has
occurred.
There are two possible situations
1) by using *8
2) by using *8123 (to pickup extension 123 when it is ringing)
The manager event looks like:
Event: Pickup
Privilege: call,all
Channel: SIP/ast163-000c
On 25/10/2009, Matt mhop...@gmail.com wrote:
This is a test... I am being told I am subscribed, but I am not getting
messages.
Gmail always seems to hide receipt of your own messages to mailing lists...
Andrew
--
Linux supports the notion of a command line or a shell for the same
reason that
This is a test... I am being told I am subscribed, but I am not getting
messages.
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On Oct 24, 2009, at 8:33 PM, Matt mhop...@gmail.com wrote:
This is a test... I am being told I am subscribed, but I am not
getting messages.
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Hello!
Thank you for the information. Regarding using the sip show peers
command, I remember somewhere seeing that it only works for static sip
accounts and does not list accounts that are dynamically stored in a
database. Most of my accounts are database entries, so would the sip
show peers
Elliot Murdock schrieb:
Regarding using the sip show peers
command, I remember somewhere seeing that it only works for static sip
accounts and does not list accounts that are dynamically stored in a
database. Most of my accounts are database entries, so would the sip
show peers command work?
Hello!
I am looking for a way to test if a SIP device is still alive or not.
I want to add this functionality in an AGI or independent script in
order ensure all the SIP phones are properly connected to the system.
Thank you,
Elliot
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Elliot Murdock schrieb:
I am looking for a way to test if a SIP device is still alive or not.
What about qualify=yes in sip.conf?
I want to add this functionality in an AGI or independent script in
order ensure all the SIP phones are properly connected to the system.
Philipp Kempgen
--
Hello Philipp,
Thank you.
I could set that up, but is that status (of qualifying) stored
anywhere (besides the log files) that a script could use?
Regards,
Elliot
On Thu, Jul 23, 2009 at 12:47 PM, Philipp
Kempgenphilipp.kemp...@amooma.de wrote:
Elliot Murdock schrieb:
I am looking for a way
Elliot Murdock schrieb:
I could set that up, but is that status (of qualifying) stored
anywhere (besides the log files) that a script could use?
You could have a script execute
asterisk -rx 'sip show peers'
and read the status for each peer.
On Thu, Jul 23, 2009 at 12:47 PM, Philipp
Hi
You can retrieve it in real time using the AMI from a script
http://www.voip-info.org/wiki/view/Asterisk+manager+API
Ish
Elliot Murdock wrote:
Hello Philipp,
Thank you.
I could set that up, but is that status (of qualifying) stored
anywhere (besides the log files) that a script could
Had an inbound email server issue, just double checking it is working
again.
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any
Yes, its working :)
Jai Rangi
ww.didforsale.com
On Thu, Apr 30, 2009 at 12:12 PM, James A. Shigley
j...@answeringserv.comwrote:
Had an inbound email server issue, just double checking it is working
again.
James Shigley
*Monroe Telephone Answering Service*
409-981-9213**
Infinity
I have an asterisk server at home. I'd like to test one just installed
elsewhere.
Both servers are behind firewalls. I can see the session start in CLI, my
congratulations is apparently playing and RTP is being sent.
Hearing no audio. Can send key presses and see audio playing changed. Peer
On Mon, 16 Mar 2009 23:00:32 -0700
Michael Higgins li...@evolone.org wrote:
I have an asterisk server at home. I'd like to test one just
installed elsewhere.
And did succeed just after emailing, of course. :(
Sorry for the noise!
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test
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On Tue, Nov 18, 2008 at 02:23:56PM +0800, lizhong zhu wrote:
hello, all of users:
after dig the code, i found that dahdi wcb4xxp is only for digium B410P. it
has VPM hardware based echo cancellation, which Junghans and openvox bri
cards do not have. anyone can tell me how to disable the
hello, all of users:
after dig the code, i found that dahdi wcb4xxp is only for digium B410P. it has
VPM hardware based echo cancellation, which Junghans and openvox bri cards do
not have. anyone can tell me how to disable the ec_write methond to support
other HFC BRI cards?
regards!
zhu
On Wed, Nov 12, 2008 at 05:55:29PM +0800, lizhong zhu wrote:
the dmesg shows:
wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got
back 0x01
printk: 13709 messages suppressed.
wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got
back 0x01
hello:
thanks for Tzafrir Cohen for dahdi testing.
I installed dahdi-2.1-r3c svn code and asterisk1-6
for testing OpenVox B400P and junghans card. i fund that there is bug (i think)
to dectect NT or TE mode. actually on the board,
i set it as TE mode, but after start wcb4xxp, but
it show the
hello, users:
I tried to change to hardhdlc in system. but i still can not make calls. the
port 4 led still can be be on.
=system.conf==
# Autogenerated by ./dahdi_genconf on Wed Nov 12 19:22:36 2008 -- do not hand
edit
# Dahdi Configuration File
#
# This file is parsed
If you have some time, interest and desire, I would like to see how
FreeSwitch compares to the 9 calls per second lost SIP message issue.
On Sun, Sep 28, 2008 at 2:55 AM, Gnu Devel [EMAIL PROTECTED] wrote:
I'm using Sipp to load test, but it lost some SIP message when I
increment Call Per
the admin
Obviously hammer can't do that
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, September 30, 2008 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] test call
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, September 30, 2008 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] test call generator
If you have some time, interest and desire, I would
Sipp looks pretty good! I don't know how I missed this one. This would've
saved me tons of time a couple months ago.
I plan on using it to load test using 2 Asterisk servers, one to initiate
the SIP calls, the other to receive. Thanks for the tip Alex.
Zac Wolfe
Safi Systems LLC
I'm using Sipp to load test, but it lost some SIP message when I
increment Call Per Second more than 9.
Regards
Grey Man escribió:
I've used both the Hammer Call Analyzer software and als to the Hammer
XMS system which is a server that they install in your rack to do the
packet captures and
Hello everyone
I am trying to look for a free test call generator that will get me some
stats like PDD, ASR and call quality etc on each route. As well as do test
at every interval too
If you know something like this please enlighten me.
Sam
Are you looking for inbound or outbound.
I can get you free inbound test DID. LMK
Jai
www.didforesale.com
On 9/27/08, Sam Tam [EMAIL PROTECTED] wrote:
Hello everyone
I am trying to look for a free test call generator that will get me some
stats like PDD, ASR and call quality etc on each
Sam Tam wrote:
Hello everyone
I am trying to look for a free test call generator that will get me some
stats like PDD, ASR and call quality etc on each route. As well as do
test at every interval too
If you know something like this please enlighten me.
Sam
On Sat, Sep 27, 2008 at 4:38 AM, Igor Hernandez [EMAIL PROTECTED] wrote:
Sam Tam wrote:
Hello everyone
I am trying to look for a free test call generator that will get me some
stats like PDD, ASR and call quality etc on each route. As well as do
test at every interval too
If
What you are looking for is SIPP: http://sipp.sourceforge.net/
It won't intrinsically tell you anything about the data; it's up to you
to appropriate the findings. But it accomplishes the generation of
traffic (and dummy media!) on a technical level.
Igor Hernandez wrote:
Sam Tam wrote:
You actually using that steve?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Saturday, September 27, 2008 6:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] test call generator
Unforunately it is outbound
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jai Rangi
Sent: Saturday, September 27, 2008 4:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] test call generator
Are you looking
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Saturday, September 27, 2008 6:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] test call generator
On Sat, Sep 27, 2008 at 4:38 AM, Igor Hernandez
I've used both the Hammer Call Analyzer software and als to the Hammer
XMS system which is a server that they install in your rack to do the
packet captures and provide you with all sorts of statistics.
I suspect the Empirix Hammer products would be able to take care of
any load, monitoring or
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sorry
just a testmail to the list, becausemy last mail does not show up on the list.
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Just a test, please discard
Looks like something is eating my messages on their way :-(
Martin
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My posts were not going thru, so I testing and debugging why.
Please ignore
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checking wheather my mail goes to asterisk users mailling list or not
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On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED] wrote:
checking wheather my mail goes to asterisk users mailling list or not
ACK.
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Thanks,
Joel
On Tue, Feb 26, 2008 at 10:47 PM, Erik Anderson [EMAIL PROTECTED] wrote:
On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED]
wrote:
checking wheather my mail goes to asterisk users mailling list or not
ACK.
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Ian wrote:
Just testing to see if my emails to this mailing list gets through.
Tried posting a question, but it failed
Thanks
Ian
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To
Just testing to see if my emails to this mailing list gets through.
Tried posting a question, but it failed
Thanks
Ian
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Sorry,
but it seems that I have banned from list.
I can reciveve, but can not send posts.
Hi!
When I use Dial(type/identifier, timeout, A(some_file))
CDR billsec starts when announcement ends. But I have to bill from when
called party answers to phone.
How can I solve my problem?
--
Suich
I've been complaining about this problem recently, but nothing has been done
about it.
I'm guessing some spam filtering software has gone badly wrong. The filtering
seems to be based on the content of the message rather than the sender.
On Wed, Nov 28, 2007 at 03:02:11PM +0100, Suity Zsolt
On 11/28/07, Jesse Molina [EMAIL PROTECTED] wrote:
I've been complaining about this problem recently, but nothing has been done
about it.
I'm guessing some spam filtering software has gone badly wrong. The
filtering seems to be based on the content of the message rather than the
sender.
Test
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test list not working
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Ok, so I was fooled :P
On 8/12/07, Stephen Bosch [EMAIL PROTECTED] wrote:
C F wrote:
OMG, someone thought that it's for real. Wow.
I don't think so. Read the sentence carefully:
On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote:
C F wrote:
No you cant. This message is being dropped as
No you cant. This message is being dropped as well.
On 8/10/07, Trevor Peirce [EMAIL PROTECTED] wrote:
C F wrote:
This is the postmaster at the list and I am notifying you that your
message failed.
Over the past two days my new posts seem to have silently been dropped.
I wonder if I can
C F wrote:
No you cant. This message is being dropped as well.
Shame. Seriously though I posted a new thread right after I posted that
reply. The reply showed up but the new thread still seems to be MIA. No
bounce or anything (and I have no filtering on this account). Weird...
Maybe I'll
OMG, someone thought that it's for real. Wow.
On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote:
C F wrote:
No you cant. This message is being dropped as well.
Shame. Seriously though I posted a new thread right after I posted that
reply. The reply showed up but the new thread still seems
C F wrote:
OMG, someone thought that it's for real. Wow.
I don't think so. Read the sentence carefully:
On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote:
C F wrote:
No you cant. This message is being dropped as well.
Shame. Seriously though I posted a new thread right after I posted that
C F wrote:
This is the postmaster at the list and I am notifying you that your
message failed.
Over the past two days my new posts seem to have silently been dropped.
I wonder if I can reply to an existing thread...
--
Does your Canadian VoIP service need CRTC-compliant 9-1-1 services?
test only. good luck!
james.zhu
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This is the postmaster at the list and I am notifying you that your
message failed.
On 8/7/07, zhu lizhong [EMAIL PROTECTED] wrote:
test only. good luck!
james.zhu
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Is the list up? I haven't gotten mail in the last 24 hours.
Alex
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Alex Roston wrote:
Is the list up? I haven't gotten mail in the last 24 hours.
Alex
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Sorry to clutter up the mailiing list, but I've been unable to post to this
list for the past 2 WEEKS!
My ISP's blocking SMPT from other than his own servers.
I think I've worked around it. - But if I see this message in the digest
then I know I'm okay.
Again. - Sorry for any inconvenience.
Gary
I don't know about bandwith consumption but look at sipp
(http://sipp.sourceforge.net/)
- Original Message -
From: khawla khawla
To: asterisk-users@lists.digium.com
Sent: Saturday, May 26, 2007 10:33 PM
Subject: [asterisk-users] test tools of Asterisk server
I am using
I am using Aserisk as a SIP server to interconnect differents PBX in differents
sites. I am now looking for a tool that can test the performance of this
solution: I mean is there a tool that enables me to test the capacity of this
SIP server in terms of simultaneous calls that could be
HP has a tool that is a free Open Source test tool / traffic generator
for the SIP protocol.
On 5/26/07, khawla khawla [EMAIL PROTECTED] wrote:
I am using Aserisk as a SIP server to interconnect differents PBX in
differents sites. I am now looking for a tool that can test the performance
of
HP's tool can be found at sipp.sf.net. Im unshure if you have to use
unstable to get rtp support or if they hasve released it as stable.
/M
Andrew Joakimsen wrote:
HP has a tool that is a free Open Source test tool / traffic generator
for the SIP protocol.
On 5/26/07, khawla khawla [EMAIL
, 2007 7:54 PM
Subject: Re: [asterisk-users] Test
Failed
On 4/26/07, gc [EMAIL PROTECTED] wrote:
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Test emails and out of office emails make my day.
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, May 01, 2007 5:37 PM
Subject: Re: [asterisk-users] Test
where
I love these :)
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 27, 2007 7:54 PM
Subject: Re: [asterisk-users] Test
Failed
On 4/26/07, gc [EMAIL PROTECTED] wrote
Failed
On 4/26/07, gc [EMAIL PROTECTED] wrote:
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-users] test
ggcc
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Sent: Wednesday, April 25, 2007 9:57:54 AM (GMT-0600) America/Chicago
Subject: RE: [asterisk-users] test
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: [asterisk-users] test
Ha
This does not directly relate, but I have NO respect for
people who use braindumps. Learn the material, do not be a
paper certification name here.
Just my 2 cents, sorry, had to get that out. :P
Cheers,
Bkruse
- Original Message -
From: Steve
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