Displaying 20 results from an estimated 4000 matches similar to: "Outbound Dialer, Agent Login and Logout"
2004 May 10
6
SIP calls-per-second performance test tool
http://sipp.sourceforge.net/
Anyone care to throw this at Asterisk to see what happens? I would,
but I am having significant temporal shortfalls recently due to the
apparent warping of the space/time continuum when I answer the phone
with clients/associates. It seems that entire days pass by before I
hang up... very odd, and very counter-productive to getting good
Asterisk work done.
JT
2013 May 20
1
Stress testing Asterisk
Hi,
I just installed Sipp 3.3?on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.
SIpp output:
----------------------------- Statistics Screen ------- [1-9]: Change Screen --
? Start Time???????????? | 2013-05-20?22:53:08:637?1369083188.637273???????????
? Last Reset
2018 Oct 04
3
CURL to post application/json
We tried to use the CURL fn to POST json, but it's sent as form data and
there seems no support for changing the Content-Type header. We switched to
invoking curl in the shell.
All the documentation I could find says there is just one parameter for the
url and an optional second for POST body. Is there an undocumented way to
set Content-Type?
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An HTML
2007 Jan 23
2
stress-test realtime voicemail with sipp
We are in the process of implementing realtime voicemail. I was wanting
to "stress-test" the system to see if or when it would fall over.
Is it possible to use sipp to create say 250 calls, each of which leaves
a message in the voicemail ?
My dialplan is currently
[default]
exten => stress,1,Answer()
exten => stress,2(vm),Voicemail(7777|su)
exten => stress,3,Hangup()
2005 Feb 07
3
SIPP load testing - unexpected message - anyone using sipp sucessfully ?
Hi,
I'd like to test Asterisk performance under more concurrent sip calls. I use
Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if anyone
is using sipp succesfully with Asterisk and is willing to share more info
about his solution ...
Any other convenient way to load test Asterisk ? Is sipp the right tool ?
Thanks in advance,
regards,
Rob.
sipp: The
2010 Sep 15
2
Digest Username/auth name mismatch
Hi
I'm sorry.
I mailed the same question again.
because, it cannot be yet solved.
any ideas with asterisk?
[Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have <aaaa>, digest has aaaa at 192.168.0.1[Aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479 handle_request_register: Registration from 'aaaa <sip:aaaa at 192.168.0.1>' failed for
2005 Jun 28
4
Anyone using SipP to produce RTP load?
Hey gang,
I've been able to use sipp to produce some call volume on our asterisk
server. The server has no problems handling 50 simul calls. But then again,
no RTP is being done. I tried to use the rtp echo ability of sipp but that
doesn't seem to work right.
I also setup a fake number in asterisk that when called by sipp, would dial
another number via PRI, hoping that some 729
2013 Mar 23
5
Optimizing Asterisk Environment
Hello Everyone,
We are getting some rather poor results (relative) with our Asterisk
setup. Not sure if we are using the sipp correctly etc.. but
nevertheless, is there any documentation that describes how we can get
the most our of our Asterisk box. For example when we hit the "too
many file" error, and fixing it using ulimit..... Also, is there any
way we can allocate sufficient
2018 Sep 29
2
WebRTC as Softphone substitute ?
Hi Olivior,
We have recently worked on a WebRTC based agent panel. As based on my
experience I think that WebRTC based phones are far better and cheaper then
those soft / sip phone. the big plus is that they are easy to customize and
developer can use the power of browser and web to build / offer features
which are not possible with regular phones.
Regarding your concern about BLF or call
2011 May 05
4
SIP secruity: username and password
Hi All;
When the endpoint register on Asterisk or initiate a call, so they exchange the sip username and password. What is the possibility that this will be capture by the hacker and how to avoid this problem?
Regards
Bilal
2016 Jun 14
4
Pet project: one step Asterisk compile on Centos 7
Hi all,
I thought I'd share I script I made (based on some of Leif's works)
that lets you download, compile and install Asterisk all in one go;
and then removed the dev tools used.
We use it quite a bit to provision systems using Ansible, but it is
easier than remembering everything every time even if you are using a
shell.
At the moment I have scripts for Centos 7 and Asterisk 13, but
2018 Oct 03
2
WebRTC as Softphone substitute ?
@Olivior
I agree that seting up WebRTC is hard, however when done it is smooth to
use. For replication you can build RPMs with working configurations.
Regarding stability, it is not being used widly, so can't say it is mature.
However we have no complain so far regarding audio or connectivity.
sometime we provide support for "allow media / mic" type issues, but you
know it is
2011 Nov 22
1
Asterisk refuses INVITE (401) and I don't know why
Hello list,
this is the communication between an Aastra 5000 PBX and Asterisk, where
the Aastra makes a call to Asterisk. For some reason, Asterisk responds
with 401-Unauthorized and I don't know why.
Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT
with this Aastra.
A1.A1.A1.A1 = IP-address Asterisk PBX
AS.AS.AS.AS = IP-address Aastra PBX
Aastra PBX makes a call
2003 Nov 27
4
Multi-line TTS Outbound Dialer
Hello,
I've been lurking around the mailing list and browsing around on
Asterisk-related links while I wait for my X100P to come in the mail.
So far I haven't found very much information related to what I want to
do with Asterisk. I was wondering if someone could point me in the
direction of any work that may already have been done on a project
similar to the one I'm trying to
2006 Nov 15
1
Attempting native bridge of
I have the following scenario:
g729 gsm
UAS <-----------> * <-----------> UAC
I am using sipp to generate the calls between the UAC and the UAS and
sending some rtp from the UAC, I want * to do transcoding but as I see
it is not. As long as I know 'Attempting native bridge' means only
passing-through the rtp ?Am I wrong?
The UAC and UAS are
2018 Sep 26
2
WebRTC as Softphone substitute ?
On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote:
>
> On 9/26/2018 4:46 AM, Olivier wrote:
>
> > Hello,
> >
> > This morning, I asked myself if WebRTC could be a viable alternative
> > to softphone deployment.
> >
> > For me, main issue with Softphones is the amount of work needed for
> > installation and
2018 Dec 19
2
New features released in ICTBroadcast
Following new features are now supported by asterisk based
telemarketing software
Auto subscription / registration after call recipient press a key in voice
broadcasting
https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer
There will be restriction to call a number in off time accordingly to
timezone of
2012 Aug 05
3
Voice Mail beep / tone detection
Though asterisk support AMD which is based on silence detection but I did
not found support of tone / beep detection in asterisk to record a voice
message for answering machines after detecting tone
Will appreciate any help in this regard
Best Regards
*Tahir Almas*
Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
Unified Communication Telemarketing
2011 Mar 30
1
dtmf_2833_1.pcap: what PCM codec? ulaw or alaw?
Hi everybody,
got it from svn:
dtmf_2833_1.pcap
*/asterisk/trunk/tests/rfc2833_dtmf_detect/configs/extensions.conf PRE-CREATION
*>* /asterisk/trunk/tests/rfc2833_dtmf_detect/configs/sip.conf PRE-CREATION
*>* /asterisk/trunk/tests/rfc2833_dtmf_detect/run-test PRE-CREATION
*>* /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.pcap UNKNOWN
*>*
2015 Mar 16
1
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan
<sonny.rajagopalan at gmail.com> wrote:
> George,
>
> I have the detailed log below. (Resending after trimming the email to 40KB.)
>
> The sequence below just repeats ad-nauseam. Is this a SIP trunk issue?
>
> Thanks!
>
I don't see anything obvious. The registration works though, right?
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