On Monday 28 January 2008 14:00:27 Rahul Yadav wrote:
Hi all
i am getting a serious problem.I am using asterisk 1.4.15 and dialing
outbound through sip.
The problem is that whenever i dial a number the other person can hear my
voice but i dont hear anything.
Have you tried:
On Wednesday 09 January 2008 09:54:59 Yves Räber wrote:
Hello,
I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have
some trouble with the CDR userfield that is not changed when using the
SET command in the realtime dialplan.
In my dialplan (extensions.conf, the file) I'm
I have a problem. I have tried everything that is in the book The
Future of Telephony as well as on the FWD (freeworlddialup) website,
and there is still a problem. My asterisk box is not able to associate
with the FWD server. I get:
Registration Rejected by [insert IP], and I can't use
On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote:
Hi,
I have the following problem that when asterisk receives SIP response 302
it cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan
On Saturday 05 January 2008 00:45:00 ameel wrote:
Is there a way to disable the b2bua feature in asterisk.
I would like asterisk to work as a sip server and not be involved in the
RTP path between phones.
You can do it by setting BOTH peers
canreinvite=yes
On Thursday 03 January 2008 15:28:15 Remco Barendse wrote:
I have an analog GSM Gateway that is connected to a normal SIP ATA device.
Basically what it does is this : when you call the extension nr. of the
SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
dial tone,
On Thursday 03 January 2008 22:15:07 William Herrera wrote:
I installed one last week (downloaded and installed the latest) and
everything went beautiful and every thing works fine, however, my client
has voice mail and no matter what phone I use, or what password I enter, or
in which way I
On Thursday 03 January 2008 16:38:35 Remco Barendse wrote:
On Thu, 3 Jan 2008, Benchev wrote:
Basically Grandstream HT286 is a single port FXS ATA.
In order to interconnect GSM gateway one would need FXO.
Are you sure it gives you new dialing tone or this is the * itself
you hear?
Yes
help would be appreciated.
By experience, after every query sequence you should use
MYSQL(Clear ${resultid}) otherwise they'll become 400.
As it is said into app_addon_sql_mysql.c
Frees memory and datastructures associated with result
set.
Hope that helps,
Benchev
-
БТК
\
lastcall=${LASTCALL}\,totalcalls=${TOTALCALLS}+1\,currentcalls={CURRENTCALLS}+1\
WHERE\ dnis=\'${IVR-Exten}\'\ AND\
ani=\'${CALLERID(number)}\')
Benchev
-
Най-добрият начин да научаваш новините,
които те интересуват.Бързо лесно и безплатно!
новини с филтър
http
,
Benchev
-
Най-добрият начин да научаваш новините,
които те интересуват.Бързо лесно и безплатно!
новини с филтър
http://www.radar.bg/
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in case they'd forgot to
disable CF, starts ringing 123456.
Hope the above will give you even more ideas.
Benchev
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Note: remember to either backslash your commas in extensions.conf or quote the
entire argument, since Set can take multiple arguments itself.
So probably with 1.4 will come.
Benchev
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asterisk-users
On Thursday 27 July 2006 15:07, Andrea Spadaccini wrote:
Ciao Benchev,
Also register= can be done only from a .conf file.
Well, I'm experimenting right now with this, and I can tell you that
register = works even with static realtime.
Not even, it *must* work because if one uses
realtime
and
friends sip or iax2 info is being read on the fly. The appropriate
extensions though, must be
addressed with switch = Realtime statement from extensions.conf.
Since all .conf files exist they have precedence. Also register= can be done
only from a .conf file.
Hope it helps.
Benchev
is complete mess with the commas ...
Benchev
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: PAP2 is not.
Benchev
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On Tuesday 04 July 2006 17:32, Martin Joseph wrote:
Who says nufone is dead?
I use them, but my did is through sellvoip.net
I have tollfree numbers with Nufone working OK.
But what I like most is the regular numbers with charge/month
but no charge/min on incoming calls...
as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press * - VoicemailMain
You should change it or use the sample from the original
extensions.conf to answer your needs.
The macro does not need to be included in a context.
And as you see priorities jump n+101.
Hope it helps.
Benchev
to use regexten in the sip peers?
I think [internal] is the place to declare the internal extensions
not in [incoming-from-sip] and once a call comes to enter
a macro loop.
Thanks,
Benchev
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Asterisk
On Wednesday 05 July 2006 20:00, John Kington wrote:
At 09:29 AM 7/5/2006 +0300, you wrote:
I have tollfree numbers with Nufone working OK.
But what I like most is the regular numbers with charge/month
but no charge/min on incoming calls...
Did your tollfree number(s) with Nufone get cut-off
it.
Benchev
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Edno utochnenie...
Ako ne sym bulgarska firma
i kupja DID ot vas shte polucha li
Invoice za BG DIDs origination?
nikakaw! nie imame nyakolko stotin klienta izwan BG. Mojete da poglednete
http://bgnumber.info za podrobna informaciya.
- Original Message -
From: Benchev [EMAIL PROTECTED
. The only
Urgent in your case is that you urgently need to
go to the wiki and have a long reading.
Please, show some efforts, benevolence or something...
Benchev
P.S.
http://www.voip-info.org/wiki/view/Asterisk+Configurations+for+connecting+with+VOIP+providers
,Hangup
Put context=internal or default in all your sip friends.
Hope that would do.
Benchev
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On Thursday 18 May 2006 16:32, Ralph Liebessohn wrote:
On 5/18/06, Benchev [EMAIL PROTECTED] wrote:
I'm trying to start with Asterisk, but I could not put 2 softphones to
talk. The asterisk server rejects the connections always when I dial.
May 17 07:49:22 NOTICE[1924]: Rejected
fine. As I did
mention before, me personally, I have one from the older models
( 5 months now) and it is rock solid.
Thanks very much for your concern.
Cheers,
Benchev
On Tuesday 09 May 2006 13:58, adibar wrote:
Hi Benchev
Mine is working now. It was set to a courious billing mode.
After
by 66.199.240.2: No authority found
My registration goes through OK.
My dial plan:
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]
Try
exten = _1NXXNXX,2,Dial,IAX2/plainvoip/${EXTEN}
or
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Benchev
Hi Adibar,
Thank you. I have also received some
recipes from Sam.
I have sent the codes to the client
and waiting for him to confirm the good news.
It is amazing sometimes how slow some things happen.
I'll let you know.
Regards,
Benchev
Hi Benchev
Mine is working now. It was set
);
Cheers,
Benchev
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for a pin and phone waits for you on w to push Enter
for the last string.
After all that you would here:Please enter the number you wish to dial...
Hope, this helps.
Benchev
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Hi Ronald,
Small mistake, see bellow:
Benchev
Just to give you an idea
I would suggest you to make two .agi files:
astcc.agi and astcc-disa.agi
In astcc.agi you'd leave everithing as it is, and enable
PIN =YES through the astcc-admin.cgi.
Thus you could dial without interogation:
exten
:-)
Benchev
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OF CHARGE EXCHANGE of the useless GSM boxes
or do immediate refund or something.
Let's see...
Cheers,
Benchev
On Wednesday 26 April 2006 01:17, adibar wrote:
Hi Benchev
News from the front. Sam is kinda offering me an exchange
of my box. But I should return it to him at my cost ;-)
Last word
recently.
* Drop: The number of packets we've purposely dropped (to lower latency).
* OOO: The number of packets we've received out-of-order
* Kpkts: The number of packets we've received / 1000.
...
Benchev
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the amount of calls on the called
channel but also on the calling channel.
In this case you should not need Answer.
Benchev
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and start making the noises.
Hope Sam could solve the problem with the factory or
exchange the goods with working ones.
Benchev
Outch... Four of them and not working... That hurts.
How do you connect them to * ? As I'm using only one
for me an X100P-FXO is sufficiant and seems to work as
good
Hi Adibar,
Any success with the gsm gateway?
I have exactly the same problem with units received this month.
The codes given by Sam are not working...
Please, let me know if you have discovered something.
Thanks in advance,
Benchev
But these are the wrong instructions again. Same as those
ones
into the extensions_table a set, which context corresponds
to where the switchis, and this is read in realtime without the need of
reloading.
Pretty much that's all I can do to help. Sorry.
Benchev
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approach, also no changing the extension.conf:
[ever_changing_dids]
#include includes/ever_changing_dids.conf
ever_changing_dids.conf
exten =
9876543210,1,Dial(SIP/user1SIP/user2SIP/user3SIP/user4SIP/user8SIP/user12|
20)
etc...
However this requires *CLI reload
Hope I've guessed right.
Benchev
) and if left off, RealTime will use the
current context, in this case sipregistrations.
Means:
[sipregistrations]
switch = Realtime/@realtime_ext ;realtime_ext or whatever the table name is
Ok i'am guessing sans voir here since I don't understand why so many
contexts are needed?
Hope it helps,
Benchev
appears the prefix 2006234500254.
Would you try:
exten=_2006234500254.,2,Set(destination = ${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/[EMAIL PROTECTED],60,tr)
Benchev
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On Tuesday 14 March 2006 17:15, Benchev wrote:
i'm trying without success to change the dst (destination) entry of the
cdr. I'm using the following:
exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
I want to record
, but in the cdr
appears the prefix 2006234500254.
But this should do it:
exten=_2006234500254.,2,Set(CDR(dst) = ${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
Benchev
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:
[sipregistrations]
But first I would try to add a field regcontext along with regexten(which
already there) in sip_users table since for the trick to work ...
read http://www.voip-info.org/wiki-Asterisk+sip+regcontext
Hope this will give you a clue.
Benchev
with the result of a call(see my previous email ), which
is bigger pain than H.
Sorry misunderstanding you.
Benchev
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} =
CONGESTION | ${RESULT} = ]?500:3)
exten = h,3,Do_things
exten = h,500,Congestion
benchev
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Is there any way not using group count, to limit calls received by every
endpoint SIP?..
Outgointlimit and Incominglimit seems to be deprecated on 1.2.x branch.
Is there another command to do that?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
-Talk kind of thing
with the exception that the handset is imbedded, so pretty much
no need of a manual.
Is your Grandstream a HT-488? If so you might be able to
simulate the spa3000 case.
Please, let me know what happened.
Best regards,
Benchev
/var/lib/asterisk/agi-bin/a2billing.php
benchev
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, but not on the Sipura's PSTN line ...
Thanks,
benchev
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