Stop advertising.
Le 26 sept. 2010 09:46, Faisal Hanif fai...@vopium.com a écrit :
Hi Abdul-Basit,
If you need only different intervals of billing you can easily do it
using any AGI as we are doing it in Perl AGIs using post call billing.
But if you need realtime billing then the most stable
Have you installed dahdi ?
And do not mix 1.4 with 1.6
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It does not support T.38 is that correct ?
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Have you tryed to generate .call files at once ?
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2009/10/14 B.Masoud @ SH i...@saudihome.com
Is there a ready add-on to asterisk that will display the ACD/ASR per
channel, source destination?
Thanks.
You can calculate by yourself with cdr's, its only statistics.
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Allmost your solutions require second server or some hardware, why do you
use shorewall ? Its a iptables rule generator with a friendly config files.
Mine was up and running in 30 min or reading some docs. And you can trace
all traffic live.
Good day.
2009/6/24 Senad Jordanovic se...@bicom.us
Jay Fenton wrote:
[ Optimised G.729A 'Howlet' for Asterisk FreSWITCH ]
Howler Technologies are proud to announce today the launch of
their fully indemnified and highly optimised G.729A solution
for Asterisk, including a unique floating license
2009/6/17 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi,
I have 2 trunks connected to my asterisk installation.
One is a inbound connection between ericsson pbx and the other is thru a
voip service.
I am using 4 digit numbers both in ericsson and asterisk..
And also i have full real prefix
Make sure you are actually setting it as:
Set(CALLERID(num)=290)
The previous poster has the formatting incorrect. If your callerID is a 4
digit
number, and you want to modify it to have the prefix on it before you send
it
back out, you can do:
2009/6/11 Olivier oza-4...@myamail.com
Hi,
I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new
Aastra SIP phones can be auto-provisioned when config files are stored in a
specific TFTP subdirectory instead of TFTP root directory.
For instance, TFTP root directory is
Remember that the time between the two digits is VERY short. You must
press
those two digits in quick succession or else the requested feature code
will
not activate.
-
Or set featuredigittimeout longer.
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2009/5/29 김무성 ki...@infosec.co.kr
I wanna connect proxy server.
my IP Phone - my asterisk - service provider's proxy server - extern
PSTN phone
but asterisk server can't register to proxy server.
I think that configuration is right.
When asterisk send to register request, proxy
2009/5/9 Don E. Wisdom d...@engineeringinc.com
I work on the salmon river in Idaho as a computer/radio tech.
All of the satellite isp's do not have the upstream capability.
Skype barely works. (you have to try upwards of 20 times for it to work)
If I have to make phone calls when I am there I
Forgot to add, it is no so bad, i mean if you are in situation where local
telco male you pay hell of a price. Or if you are in location not covered by
any telco, i would go by sattelite option.
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Not a taboo at all, you are providing your knowledge to setup the call
center for example, and i your support in future. It is commen practice.
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To
2009/5/9 Dean Collins d...@cognation.net
Perfect office rackmount asterisk server?
http://www.tgdaily.com/html_tmp/content-view-42372-135.html
Lacking dual hdd for raid 1.
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2009/5/9 Steve Edwards asterisk@sedwards.com
On Fri, 8 May 2009, Dave Walker wrote:
I have a question for those who have done a few professional installs of
Asterisk. Is it taboo to use something like AsteriskNow/FreePBX/Trixbox
to get a base installation of Asterisk installed and
2009/5/7 Jim Dickenson dicken...@cfmc.com
I have * 1.6.0.9 with dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.2, libpri
1.4.10 and wanpipe-3.4.1 running on CentOS 5.3 64bit.
I have 2 ports of the a104d configured for use with PRI lines and 2 ports
configured for use with Adtran Total Access 850
2009/5/7 Jim Dickenson dicken...@cfmc.com
*From: *Grygoriy Dobrovolskyy megaho...@gmail.com
*Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Date: *Thu, 7 May 2009 12:20:07 +0200
*To: *Asterisk Users Mailing List - Non-Commercial
2009/4/20 Gary Li garyli0...@gmail.com
Hi,
I had some experience on Asterisk 1.0.7 and 1.2.0.
Now, I want to do something on the New Release of Asterisk 1.6.xx.
I want to know wheather there are already web GUI for use now in the
release.
Or still nedd integrate some other third part
On the last page
http://qvlweb.blogspot.com/2009/04/asterisk-pbx-system-install-04-pbx-test.html
there is a small screen, number 3 from bottom, looks like you are editing
exgensions.conf not extensions.conf.
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2009/4/1 Michael mich...@networkstuff.co.nz
haw haw haw...
April Fools Day is over in this part of the world.
Hey dont kill the magic ! :)
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To
2009/4/1 Shaun Wingrin voi...@gmail.com
Hi,
Why does this warning occur and what are the implications of it? I'm
concerned about calls never getting hung up.!
chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to
call
2009/3/27 Marco Sambo derwid...@gmail.com
I have to try Skip2PBX, integrated into my Asterisk machine, but it seem
more invasive than Gizmo5 opensky. Doesn't it?
Marco
Skip2pbx is based on freebsd so i dont think thank you can install it on the
same pc.
2009/3/27 Mr. James W. Laferriere bab...@baby-dragons.com
Hello Mark Miquel ,
On Thu, 26 Mar 2009, Mark Michelson wrote:
Miguel Molina wrote:
Hi all,
For those of you people that use Agents (with Agentlogin, not
AgentCallbackLogin) on a call center, I have this need: when the
2009/3/26 Alejandro Cabrera Obed aco1...@gmail.com
Dear all, I've read some news about Sisky
(http://www.yeastar.com/Products/SiSkyEE.asp), a service to
interconnect Skype clients with SIP clients.
Does anybody test Sisky and can tell me about his experience ???
(Sisky runs on Windows
skip2pbx is the best i tryed, but nasty price ;)
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2009/3/24 Christian Victor christ...@victormedia.de
Hi!
A customer of mine wants to connect an asterisk system with 240 to 480
lines to a PSTN switch. To save the costs for E1 cards and the corresponding
E1 mainlines he wants to connect the system to the switch by a SIP trunk.
Phones will
Hello everybody, i am searching a solution for a videoconferencing, Any
solution (Free/commercial). Asterisk is a great software, but recently we
have more and more demands about videoconferencing of 3 or more peoples,
Existing solutions are heavy and costly, around 2500€ for 1 client. This is
2009/3/17 zoach...@securax.org zoach...@securax.org
Vincent Li wrote:
Hello,
I just had a meeting about a pilot project going on in our University,
The
project manager has done some research in the past year and concluded
that
Asterisk can not scale well to large user base like
2009/3/16 Alex Balashov abalas...@evaristesys.com
I don't know how good Asterisk's GR.303 support, but you could use DLCs as
well. However, that's a lot of complexity and (seemingly) immature
functionality liability to achieve the same end you'd get with a channel
bank. The only benefit is
2009/3/13 Andrew Thomas a...@datavox.co.uk
I think I understand what you mean now. The biggest difference between
CLI and ANI is that ANI can't be blocked/withheld (like you can with CLI
by using 141). It also uses different signalling. This is mainly used
by law enforcement agencies to
2009/3/12 Tristan tris...@telemaque.fr
Hi,
Send it to cups via the FaxDispatch script ;)
Regards,
Tristan
voip crazy a écrit :
Hello list,
I have an asterisk / hylafax / iaxmodem configured in one machine. All
is working nicely. Now I need the fax to be print when arriving.
2009/3/12 voip crazy voipcr...@gmail.com
Hello list,
I have an asterisk / hylafax / iaxmodem configured in one machine. All
is working nicely. Now I need the fax to be print when arriving.
¿Anybody have this feature implementing in their systems?
¿How is the best way to get that?
Any
2009/3/10 Ali Jawad alijaw...@gmail.com
Great Job Bogdan
On Tue, Mar 10, 2009 at 12:52 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi,
When trying to cluster Asterisk boxes to gain scalability and more
performance, there is now a new simple and efficient solution for doing
2009/3/8 Marco marcota...@libero.it
Hi List,
I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years
on my lab test setup and I appreciate it. Moreover the global quantity of
fax handled by this setup is not very high.
I'll be involved in a more complex system for a
2009/3/8 Sven Geggus use...@fuchsschwanzdomain.de
Gavin Henry gavin.he...@gmail.com wrote:
Just transfer them to your meetme extension after you've called them.
Hm, how would I do this? Until now call switching usually ended for me when
the call has been established.
I'm using a SIP
2009/3/6 BERGANZ François franc...@acropolistelecom.net
hello,
I will do a server to do a lots of conferences (MeetMe).
I want to know that if I dont use a digum card, the limit of simultaneous
calls is harder without a card than with a card ?if, yes, how harder is the
limit?
--
*发件人：* Grygoriy Dobrovolskyy
*发送时间：* 2009-03-04 16:30:06
*收件人：* Asterisk Users Mailing List - Non-Commercial Discussion
*抄送：*
*主题：* Re: [asterisk-users] after install the zaptel but the rtp failed
2009/3/4 邱磊 qiulei...@163.com
hi Grygoriy :
appreciate your reply
2009/3/4 邱磊 qiulei...@163.com
hi Grygoriy :
appreciate your reply ,
that's my cli command:
CLI zap show status
Description Alarms IRQbpviol
CRC4
ZTDUMMY/1 1 UNCONFIGUR 0 0
0
Is't all right? forward your echo
2009/3/3 邱磊 qiulei...@163.com
hi everyone:
now ,i have a strange situation: I want to make a meetme conference and
install the zaptel1.4* in my asterisk.
every things seem well but it did't work normally.
I use the Playback app for test .It didn't reply any voice.I tried in
another
2009/2/27 Alistair Cunningham acunning...@integrics.com
Ignacio,
Our Enswitch product matches all these requirements; indeed it goes well
beyond them:
- We scale far beyond 3000-4000 concurrent calls. We'd consider such a
system medium sized. At this size the system is fully
2009/3/1 michel freiha mich...@gmail.com
Dear David,
I'm using G729 pass though mode...No transcoding is used here
Regarding concurrent calls, I have 3 asterisk servers working in load
balancing mode...The issue that the same problem appear on 3 asterisk...each
asterisk handle around 150
2009/2/27 Wilton Helm wh...@compuserve.com
I assume that the relevant application requires some non-trivial CPU
power. I would
exclude e.g. a 486-based systems.
I'm not sure that's the case. The industry has gone in the direction of
throwing lots of silicon at a problem, often as an
2009/2/25 Klaus Darilion klaus.mailingli...@pernau.at
Hi!
I have a setup with Asterisk in front of a PBX connected with ISDN to
the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing
ENUM for outgoing calls and allows incoming calls per SIP.
Recently the IP connectivity for
Big companies, especially those with major computing systems use paid
software
because they want a vendor they can hold responsible for it.
As for OSS and FOSS, it is majorly used by the sort of businesses and
individuals who call me (and other IT pros) up and talk the talk, but they
2009/2/18 Asterisk Asterisk nt_aster...@yahoo.com
Thanks for the feedback. I did some research and it looks like you were
calling over international lines. It also appears that there was high than
average static on the line, which is not normal for my system. It's true
that I threw my
I think in this case when 5k call are involved i think all the difficulty of
the project is to split the load on different parts of the system. In my
case i would do it like that:
Phones ---Opensips (Double server with heartbeat and in different places)
|
2009/2/17 Danny Nicholas da...@debsinc.com
Just a laypersons opinion – I'm sure others here have better answers or
justifications.
1. no (at least not realistically, mathematically there are some)
2. perhaps – bandwidth would be your primary concern since 5K calls
would take 150 M
2009/2/13 Philipp Kempgen philipp.kemp...@amooma.de
Benny Amorsen schrieb:
Top posting is annoying. Gmail is broken; maybe I should just killfile
@gmail.com.
Emails sent through Gmail's *web interface* are broken. :-)
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany
2009/2/16 Fabio Mosti fmo...@gmail.com
Hi All,
I need to setup asterisk to receive fax.
I'm try Spandsp (opensource) and Attrafax (commercial) both on
asterisk 1.4.23) but the results are disappointing.
with spandsp many times the fax arrives cut.
with Attrafax i have some problem.
2009/2/13 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
I've been involved with getting better data for running Asterisk on
the Amazon EC2 cloud computing system. Here are some calculations
I've made on costs based on current published
2009/2/16 Michael mich...@networkstuff.co.nz
Anyone have any idea or solution (Opensource or commercial) to suggest
me
?
Best Regards
Try hylafax with IAXmodem. The best results i had it the multitech
modems
directly connected to FXS PCI card, you have a nice web
2009/2/16 Grygoriy Dobrovolskyy megaho...@gmail.com
2009/2/16 Michael mich...@networkstuff.co.nz
Anyone have any idea or solution (Opensource or commercial) to suggest
me
?
Best Regards
Try hylafax with IAXmodem. The best results i had it the multitech
modems
directly
2009/2/16 SIP s...@arcdiv.com
Grygoriy Dobrovolskyy wrote:
2009/2/13 Tzafrir Cohen tzafrir.co...@xorcom.com
mailto:tzafrir.co...@xorcom.com
On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
I've been involved with getting better data for running Asterisk
The desktop versions of snom support Openvpn, i am not sure about M3 (dect).
Take a tour to their site.
2009/2/12 Frank Bulk - iName.com frnk...@iname.com
Not in the form factor that you would expect.
Can I ask why? Most modern VoFi phones support WPA2.
Frank
-Original Message-
Hello, if you dont know iptables that much, and would like to see more user
friendly configuration method, i suggest you to use Shorewall, which is
very flexible, has some clear logs, and generates same iptable rules behind.
2009/2/8 David fire ddf...@gmail.com
denay permit are in sip.conf and
How many ports have you forwarded for the * ? (in rtp.conf)
If a limited amount (50-100), try to forward more.
2009/1/29 GNUbie gnu...@gmail.com
Hello all,
In addition to my previous e-mail, below is a more verbosed messages I
got on my Asterisk shell when calling from another GTalk User ID
note also that when I tried
calling the GTalk ID, the Asterisk box was idle or there was no any
other on-going calls.
Regards,
GNUbie
On Thu, Jan 29, 2009 at 4:15 PM, Grygoriy Dobrovolskyy
megaho...@gmail.com wrote:
How many ports have you forwarded for the * ? (in rtp.conf)
If a limited
You enabled port forwarding, but have you actually forwarded any ports ?
Defaults are
tcp 5060
udp 1-2
2009/1/29 Tamer Higazi th9...@googlemail.com
Hi people!
I am not getting smart getting asterisk 1.6 behind a NAT to run.
1. I enabled IP forwarding on debian linux
2. told
Paste your register lines (hide pass)
2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com
I want to establish a trunk SIP between Asterisk 1 and Asterisk 2, using
a sip account (Asterisk 1 acting as a conventional sip user).
Thanks
Regards
Danny Nicholas escribió:
Inter-* registry
with a 401 message (with Digest algorithm, realm and nonce).
I want to configure the Asterisk 1 in order to send REGISTER with
credentials.
Thanks
Regards
Grygoriy Dobrovolskyy escribió:
Paste your register lines (hide pass)
2009/1/29 Imanol Pardavila imanol.pardav...@ibercom.com
Disable the firewall which is enabled by default in centos
Run
system-config-securitylevel
Set both Security Level and SELinux to Disabled and hit OK:
http://images.howtoforge.com/images/perfect_server_centos_5.2/big/24.png
2009/1/25 David fire ddf...@gmail.com
paste all your sip.conf or
try lspci
2009/1/26 Tzafrir Cohen tzafrir.co...@xorcom.com
On Mon, Jan 26, 2009 at 01:45:56PM +0100, Philipp Kempgen wrote:
Tzafrir Cohen schrieb:
On Mon, Jan 26, 2009 at 05:24:03PM +0530, David @ULC wrote:
Which command to run which will auto detect all hardwares present in
the
Copy paste from freeswitch.org
Asterisk uses a modular design where a central core loads shared objects to
extend the functionality with bits of code known as modules. Modules are
used to implement specific protocols such as SIP, add applications such as
custom IVRs and tie in other external
Or boot in single user
type passwd and done.
2009/1/22 Jim Dickenson dicken...@cfmc.com
What I have done in the past to set the password for root is to boot in
rescue mode and edit /etc/shadow setting the password to some know value
from another system.
--
Jim Dickenson
try to know the whole string ?
core show channels
2009/1/16 Carlos Chavez cur...@telecomabmex.com
I have this call:
SIP/protel-525512047 default 90445528885371 1 Ringing
AppDial (Outgoing Line) 90445528885371 264:24:2
(None)
I cannot use the
Here you go http://tinyurl.com/a7tkkz
2009/1/12 chinmay chakraborty chinmay.chakrabo...@gmail.com
Hello,
I am having problems getting one xlite clients to communicate through
asterisk. I am getting an error message:
chan_sip.c:15593 handle_request_register: Registration from 'chinmay
Can you show me your script please ? For which version is it ?
2009/1/10 Markus A. Wipfler mar...@infocom.co.ug
Another way to monitor this via cacti (for example if you don't have snmp
support for asterisk or need to customize what you are graphing) is to
create a new data input method in
I wonder if the same is possible with centreon ?
Someone is using centreon here ?
2009/1/11 Markus A. Wipfler mar...@infocom.co.ug
On Jan 11, 2009, at 2:43 PM, Grygoriy Dobrovolskyy wrote:
Can you show me your script please ?
if for example you had 4 trunks then the below should give you
You should turn rtcp off in the phones settings.
2009/1/12 Rajkumar S rajkum...@gmail.com
Hi,
While looking for the cause of disturbance in call I found this error
coming in console
RTCP SR transmission error, rtcp halted
Google search only shows some bug reports relating to MOH and
2009/1/9 Steve Howes st...@geekinter.net
On 9 Jan 2009, at 16:36, Klaus Darilion wrote:
Hi!
I want to detect brute-force password hacking attacks - thus if there
are too many failed login attempts for a SIP account I want to lock
this account.
Does somebody have any ideas how this
700-800 is the maximum limit without transcoding on very optimized setup. I
would call it suicide without a failover solution. Why dont you consider the
dns srv for load balancing among 2 servers ?
2009/1/8 Scott Plante spla...@insightsys.com
Jerry, back in August you were thinking about
core show function SIPPEER
2009/1/6 Klaus Darilion klaus.mailingli...@pernau.at
since 1.4 you can also use
setvar=foo=bar
in sip.conf when configuring the peer. Then the channel variable foo is
automatically set to bar for calls initiated by this peer.
regards
klaus
Philipp Kempgen
Xorcom had something, usb bri, but it is pricey. If you dont need to change
provider and planning to stay with bri, why dont you buy another bri phone ?
2009/1/7 Matthias Apitz g...@unixarea.de
Hello,
I own one of these netbooks Asus Eee PC 900, mine is running FreeBSD 7.0,
and a
Linux
2009/1/7 Max Alex max.aster...@gmail.com
Hi,
Thanks for your reply
Can you suggest me how can we avoid it by doing any configuration changes
in asterisk.
So the freeze issue may not be occurred again!
Please provide me some help!!!
Thanks in advance!
Thanks,
Max Alex
Voip Developer
2009/1/7 TianLun Song stl...@gmail.com
From the product description, i dont think Gizmo5 allows me to register the
client with my asterisk. If i am wrong, please let me know
On Wed, Jan 7, 2009 at 4:43 PM, Rodolfo Alcazar Portillo
rodolfo.alca...@padep.org.bo wrote:
Missed the thread,
try do add
fromdomain=acme.com/sip.acme.com
fromhost=acme.com/sip.acme.com
2009/1/6 Frank Bulk frnk...@iname.com
I tried that before, but I just tried it again. Unfortunately, the same
thing:
No user '5551236049' in SIP users list
Found peer 'ACME' for '5551236049' from
Sometimes it's a problem of the timing, do you have this problem with normal
call's ?
2009/1/6 john_re john...@fastmail.us
I'm having a problem getting a good clear output sidnal from Ekiga to a
VOIP conference call using the Ekiga.net free conference call system.
I'm told that each time I
10:53:44 WARNING[5602]: channel.c:889 channel_find_locked: Warning:
Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL
Can someone tell me to what it is related ?
asterisk 1.4 freepbx
Thank you
Grygoriy Dobrovolskyy
It is very simple take openser(opensips/openser/kamalio) the openser
community is great, the project have been here and tested for a years in
production, used by the biggest companyes (millions!) of users, it's a
carrier grade soft ;) in combination of cdrtool + opensips + mediaproxy you
can get
2008/11/21 Yehavi Bourvine [EMAIL PROTECTED]
Hello,
Our university has to upgrade soon its old Nortel PBX's which holds
around 10,000 extensions tied to 5 PBXes. Up to now we thought about
commercial solutions but now there is a window openning for open source
solution. However, I need
server problem's
2008/11/21 Luis Morales [EMAIL PROTECTED]
Does any know what happens with svn repository on svn.digium.com ?
--
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
2008/11/20 Nitzan Kon [EMAIL PROTECTED]
Hello!
We're looking for a solution to reliably load balance our
Asterisk boxes. So far we've been using a hodge-podge of
directing different services to different boxes/IPs, but
eventually I'd like to consolidate things so we can present
a single IP
2. Overkill to install and maintain (if we can get a simpler
solution)
I am not agreed on point 2:
If I understood how to install opensips + heartbeat WITHOUT knowing any
program (opensips ? heartbear ?) or programming language(hell yes!) in a
week ( just knew what's invite and bye ;) a more
Use snom M3 Siemens got some problems.
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2008/10/6 Tarek Sawah [EMAIL PROTECTED]
i haven't facedthse tpe of problems you mentioned with mysql.. but there is
one thing that you need to edit the sip.conf iax.conf or you can use the
sample ones in the samples folder..
other than that.. i've been with trixbox for over three years now..
2008/10/5 Andrew Kohlsmith (lists) [EMAIL PROTECTED]
On October 3, 2008 04:15:26 pm Tariq .. wrote:
it is FRING i'm sorry for the mistype...
www.fring.com
I just downloaded it for the iphone... it's pretty cheap looking, crashes
occasionally and appears to force all audio through their
I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way behind of a
good sip provider, thay are simply not suitable for business, i hope it
would not be the case of asterisk addon. Also i wonder if skype auto relay
will be disabled (bandwith), wait and see...
2008/9/26 randulo [EMAIL PROTECTED]
Get Olle to call in for once in his life!
Mark did say IM and video, IM first. It's all gonna happen. (just not
right away)
http://lists.digium.com/mailman/listinfo/asterisk-users
Video ? that could be really nice but limited to pc/macasteriskwhatever.
2008/9/26 Kevin P. Fleming [EMAIL PROTECTED]
Brian J. Murrell wrote:
And so will this channel driver also allow Skype to use my resources
(CPU, bandwidth -- i.e. Internet for which many have usage caps, etc.)
the way the Skype client does?
The Skype engine in Skype For Asterisk does not
Yo can do it with Playtones(!440) !440 is for france seach yours in
indications.confhere is the example script from asterisk-france, the guy had
the exact same problem
[Appel_Sortant_Isdn]
exten = _0,1,Set(Flag_Playtone = 0)
exten = _0,n,Playtones(!440)
exten = _0,n(Continue),Read(Digits,,1,,,3)
2008/8/30 Shariq Khan [EMAIL PROTECTED]
When i dial out any number through PRI it gives the following error every
time, while incoming calls works fine
I have sangoma E1 PRI card.
-- Executing Dial(SIP/2000-081b9938, Zap/g0/0501125||) in new
stack
-- Requested transfer
Every one PSTN line connected to the FXS port of sipura..
Though these 4 lines comes in one cable if that has to do with anything!
Not clear for me, develop some more you topology.
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I have a simple cdr configured with the default tables, here is a row of a
good cdr report
calldate | clid | src |
dst | dcontext | channel | ect . ect
2008-08-29 10:16:49 | C. BOUTON 40 | 40 | XXX |
Remove pstn lines from sipura and call sipura to sipura ... any problems ?
Still with pstn lines removed call sipura1 sipura2 and after sipura
3sipura1 do you still hear any voices? if not it's you cable to pstn.
Give us feedback
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We had some problems with siemens 675ip with audio, but with the correct
setup they disappeared, we are using one base and 2 phones.
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AstriCon 2008 - September 22 - 25 Phoenix,
I you have such a problems with siemens you should consider 8 voip port
linksys gateway with dect bases, their gateway is rock solid and cheap.
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AstriCon 2008 - September 22 - 25
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all
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