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www.zuhair.info
*http://pk.linkedin.com/in/zuhairraza** ***
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Yes running from h
exten = _X.,1,Dial(SIP/1*100)
exten = h,1,AGI(cdr.php,11)
Regards,
Zohair Raza
On Fri, Dec 16, 2011 at 6:42 PM, Danny Nicholas da...@debsinc.com wrote:
You are running the AGI from the h() exten? Otherwise I wouldn’t expect
CDR(end) to populated or correct
Still same, even when I am trying to write in one agi and calling it using
DeadAGI
Regards,
Zohair Raza
On Fri, Dec 16, 2011 at 6:56 PM, Danny Nicholas da...@debsinc.com wrote:
Try this
exten = _X.,1,Dial(SIP/1*100)
exten = h,1,wait(10)
exten = h,n,AGI(cdr.php,11
thanks, It worked for h!
and if I want in DeadAGI? I want cdr function in the same AGI.
Regards,
Zohair Raza
On Fri, Dec 16, 2011 at 7:08 PM, Eric Wieling ewiel...@nyigc.com wrote:
From cdr.conf.sample:
; Normally, CDR's are not closed out until after all extensions are
finished
Hi,
http://blog.tmcnet.com/blog/tom-keating/asterisk/using-monit-tool-to-monitor-asterisk.asp
Regards,
Zohair Raza
On Sun, Dec 18, 2011 at 9:26 AM, virendra bhati virbh...@gmail.com wrote:
Hi List,
I have asterisk 1.6.2.20 installed at production server, I have 2 SIP voip
trunk
(calledip)=${CHANNEL(to)}) doesn't work
Regards,
Zohair Raza
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may this helps,
In cdr.conf, set endbeforehexten=yes
Regards,
Zohair Raza
On Wed, Dec 28, 2011 at 4:46 AM, Vinod Dharashive vdharash...@gmail.comwrote:
Hi team,
On event of no answer in CDR the starttime and endtime of call remains the
same.
Is there any way how can actually track call
all of them have a wiki page
http://lmgtfy.com/?q=Asterisk
http://lmgtfy.com/?q=freeswitch
http://lmgtfy.com/?q=openser
http://lmgtfy.com/?q=TrixBox
Regards,
Zohair Raza
On Tue, Jan 3, 2012 at 5:47 PM, Kaushal Shriyan kaushalshri...@gmail.comwrote:
Hi,
Please help me understand
Hi,
This may help you.
http://www.techistan.com/2010/05/31/difference-between-kamailio-and-freeswitch-or-asterisk-and-more-with-mierla/
Regards,
Zohair Raza
On Tue, Jan 3, 2012 at 5:57 PM, Kaushal Shriyan kaushalshri...@gmail.comwrote:
On Tue, Jan 3, 2012 at 7:23 PM, Zohair Raza
This works fine for me,
$dialstatus = $agi-get_variable(DIALSTATUS);
$cdr['dialstatus'] = $dialstatus['data'];
Try as it is, I believe it's because of concatenation.
Regards,
Zohair Raza
On Fri, Dec 2, 2011 at 4:27 PM, Tony Mountifield t...@softins.co.uk
Hi,
Try setting CDR(clid)
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati virbh...@gmail.com wrote:
Hi,
I am using phpagi for agi scripting. and want to update callerid number
but didn't get any success. please help me how to update PHPAGI is new for
me. Below
In phpagi
$agi-set_variable(CDR(clid) )
and to get it
$agi-get_variable(CDR(clid))
Regards,
Zohair Raza
www.zuhair.info
*http://ae.linkedin.com/in/zuhairraza** ***
On Thu, Jan 12, 2012 at 12:51 PM, virendra bhati virbh...@gmail.com wrote:
How to used it in AGI ? I think it's
Any variable can be set and get from agi
CDR(clid) is a CDR variable
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:51 PM, virendra bhati virbh...@gmail.com wrote:
How to used it in AGI ? I think it's Dialplan apps.
On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza engineerzuhairr...@gmail.com
Phpagi also has predefined method
$agi - set_callerid();
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 1:02 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
Any variable can be set and get from agi
CDR(clid) is a CDR variable
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:51 PM
and as soon as the credit goes 0,
hangup all calls for this customer.
Is there any other way to achieve this ?
Regards,
Zohair Raza
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On Wed, Jan 18, 2012 at 9:35 AM, virendra bhati virbh...@gmail.com wrote:
Hi Zohair,
By using only asterisk it's not possible. So used progremming languages
and do realtime billing at your ends.
like 1st caller will take complete amount ($5) and if 2nd call will come
then deduct
Oh yes that will be more suitable but will still need to do it via AMI
Regards,
Zohair Raza
On Wed, Jan 18, 2012 at 11:35 AM, virendra bhati virbh...@gmail.com wrote:
Batter is used DB to store intime of call then when ever currect used time
is required then deduct from intime - current
Thanks for this explanation Dany!
Regards,
Zohair Raza
On Mon, Feb 6, 2012 at 10:11 PM, Danny Nicholas da...@debsinc.com wrote:
You are mis-understanding the concept – the noanswer option is playing the
file as you requested, but since you aren’t answering the call, no channel
:
(congestion,noanswer)
-- SIP/1000-0019 Playing 'congestion.slin' (language 'en')
-- SIP/1000-0019AGI Script agi.php completed, returning 0
Regards,
Zohair Raza
On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote:
Hey Danny,
I've this thing exactly running
Sammy,
Problem is at phones, with a linksys phone it works but with eyebeam and
fanvill it doesn't
Maybe they don't support early media.
I think i will have to stick with ResetCDR and that will be okay now as
I've modified the code for that
Thank you
Regards,
Zohair Raza
On Tue, Feb 7, 2012
Yes,
Thanks
Regards,
Zohair Raza
On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind govoi...@gmail.com wrote:
Exactly that's what I expected.
Great - now have fun
On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
Sammy,
Problem is at phones, with a linksys
Confirmed as well, played back with wireshark and audio was there but phone
was ringing.
Thanks again.
Regards,
Zohair Raza
On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind govoi...@gmail.com wrote:
Hi,
Given invites seems fine, can you take a wireshark trace of the call on
your eyebeam
was for SIP switching
Regards,
Zohair Raza
On Tue, Feb 7, 2012 at 3:38 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What
technology FreeSwitch is used and asterisk don't. I don't know it's the
right or wrong
Virendra,
You can test your box with sipp
http://etel.wiki.oreilly.com/wiki/index.php/Using_SIPp_to_Stress_Test_Asterisk
I have verified my Asterisk 1.8 box handling 500 concurrent calls and 15
calls per seconds with 20% cpu, without transcoding.
Regards,
Zohair Raza
On Wed, Feb 8, 2012
It's 4 core Intel(R) Xeon(R) CPUX3220 with 6GB RAM
Regards,
Zohair Raza
On Wed, Feb 8, 2012 at 5:46 PM, Bryant Zimmerman brya...@zktech.com wrote:
Zohair
What kind of hardware spec are you running CPU, MEM, Drives?
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
add more conditions in the same way
Regards,
Zohair Raza
On Fri, Feb 17, 2012 at 1:00 PM, CDR vene...@gmail.com wrote:
My customer needs to set a forwarding based on number of rings,i.e.,
if the phone rings 5 times (user-selectable), then try another number.
Is there a way to do
Hi Kevin,
http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension
this says 4 active ports for one call
Regards,
Zohair Raza
On Wed, Feb 22, 2012 at 4:38 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/22/2012 06:26 AM, virendra bhati wrote:
Does
Try passing escape character
GET DATA $filename $timeout $max_digits $escape_character
Regards,
Zohair Raza
On Wed, Feb 22, 2012 at 6:40 PM, Chris Bagnall
aster...@lists.minotaur.ccwrote:
Greetings list,
I've done AGI scripting before, but in the past I've always wanted control
I gave it from phpagi.
It works for me using phpagi's function get_data
http://phpagi.sourceforge.net/phpagi22/api-docs/phpAGI/AGI.html
Regards,
Zohair Raza
On Wed, Feb 22, 2012 at 7:20 PM, Chris Bagnall
aster...@lists.minotaur.ccwrote:
The problem seems to be that GET DATA returns control
You want to allow single IP or whole subnet ?
Regards,
Zohair Raza
On Wed, Feb 29, 2012 at 4:44 AM, sean darcy seandar...@gmail.com wrote:
An outside device can't register:
WARNING: getnameinfo(): ai_family not supported
WARNING: chan_sip.c:14456 parse_register_contact: Domain
'69
Hi,
this can also be helpful
http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/
Regards,
Zohair Raza
On Wed, Mar 7, 2012 at 7:53 PM, Danny Nicholas da...@debsinc.com wrote:
Nothing against fail2ban but in this case I think the “route drop”
solution is more appropriate
Hi,
How to allow registered sip users to call without re-authentication
insecure =yes/very are deprecated in 1.8
I want to avoid fromuser= in peer configuration. When I add this in peer
asterisk, my asterisk accepts call otherwise it says username mismatch.
Please help
Regards,
Zohair Raza
They don't require authentication of invites which I do need
Regards,
Zohair Raza
On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini ldard...@gmail.com wrote:
2012/3/22 Zohair Raza engineerzuhairr...@gmail.com
Hi,
How to allow registered sip users to call without re-authentication
,
Zohair Raza
On Thu, Mar 22, 2012 at 3:26 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
They don't require authentication of invites which I do need
Regards,
Zohair Raza
On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini ldard...@gmail.comwrote:
2012/3/22 Zohair Raza engineerzuhairr
I've figured this out using match_auth_username =yes
Thanks
Regards,
Zohair Raza
On Thu, Mar 22, 2012 at 3:33 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
My main box is asterisk 1.8
and there are two boxes, one asterisk 1.8 and other 1.4
with 1.4, I don't need to define
videosupport=yes in sip.conf
Regards,
Zohair Raza
On Mon, Apr 9, 2012 at 12:22 PM, p070075 Muhammad Atif Ramzan
p070...@nu.edu.pk wrote:
Hi
I am new to asterisk 1.4 can someone tell about how to enable the video
conference in asterisk-gui 2.0
your destination address is not correct,
on CLI, check what is actually being passed in Dial application
Regards,
Zohair Raza
On Wed, Apr 18, 2012 at 2:04 AM, motty.cruz motty.c...@gmail.com wrote:
Hello All,
I'm gettint this error, started recently when I upgraded to 1.8.10 from
1.8.4
that variable in hangup extension but not the channel variable.
Regards,
Zohair Raza
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Hi,
http://www.voip-info.org/wiki/view/Asterisk+log+queue_log
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
Regards,
Zohair Raza
On Wed, Jun 13, 2012 at 7:38 AM, Pratik Shrestha pratik...@gmail.com wrote:
Dear All,
I am making asterisk report using CDR values given
- --
2- -
Regards,
Zohair Raza
www.zuhair.info
http://ae.linkedin.com/in/zuhairraza
On Fri, Jun 15, 2012 at 7:52 AM, Jakson Kalsson sipmaill...@gmail.com wrote:
Hi all, I have a project for the 3G related, AMR and AMR-WB support.
I'm using the client develop suite
on figuring this out please.
Thanks
Regards,
Zohair Raza
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try with SipAddHeader(uri=answer-after=0)
check syntax for Addheader
Regards,
Zohair Raza
On Fri, Jul 13, 2012 at 1:42 PM, upendra uppi...@gmail.com wrote:
Hi,
I am trying to write dial plan for sip to auto answer (auto attend) the
incoming call to the sip phone.
- If i call from sip1
In dialplan
http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader
Regards,
Zohair Raza
On Fri, Jul 13, 2012 at 1:50 PM, upendra uppi...@gmail.com wrote:
Hi,
thanks , i need to put this in the sip context...
regards
Upendra.
On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza
Also you could have a look at openfire and it's Asterisk-IM plugin
On Wed, Sep 12, 2012 at 10:41 AM, Hans Witvliet aster...@a-domani.nlwrote:
1.8 machine
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specify multiple subnets with ';' like:
permit=192.168.2.0/255.255.255.0;192.168.1.0/255.255.255.0
Regards,
Zohair Raza
On Mon, Nov 19, 2012 at 4:12 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi;
How I can make my configuration to allow the sip phones only from specific
IP addresses range
exten =520xx,1,NoOp(Caller-ID: ${CALLERID(all)})
exten =520xx,2,GotoIf($[${CALLERID(num)} = 0666XX ]?3:4)
exten = 520xx,3,Dial(SIP/224, 30)
exten = 520xx,4,hangup
Regards,
Zohair Raza
On Mon, Jan 14, 2013 at 7:43 PM, Michelle Dupuis mdup...@ocg.ca wrote:
Check out
you need to run full command, like
agi show commands topic answer
agi show commands topic gosub
agi set debug on
Regards,
Zohair Raza
On Tue, Jan 15, 2013 at 1:05 PM, Muhammad mohammad.ghaz...@gmail.comwrote:
Hi,
in CLI, I type agi show or other agi commad, but response me command
on logic in your script, you can also separate users by
contexts
* *
On Tue, Jan 15, 2013 at 12:39 PM, Zohair Raza
engineerzuhairr...@gmail.com wrote:
you need to run full command, like
agi show commands topic answer
agi show commands topic gosub
agi set debug on
Regards,
Zohair Raza
Hi,
Is it working for anyone?
I have tried with
trustrpid=yes
sendrpid=yes/pai
but can not get it working, Asterisk cli shows prevented message like this.
Connected line update to SIP/1231-0200 prevented
Regards,
Zohair Raza
Thanks for pointing that
have it disabled now
But caller id still not getting updated
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It works fine on my SPA504G
but not on 7942
Regards,
Zohair Raza
On Sat, Feb 16, 2013 at 9:32 AM, Vladimir Mikhelson v...@mikhelson.comwrote:
Zohair,
I am not sure about the specifics of 7942 as I use 7906.
Connected line CID shows up on my 7906 with the following sip.conf
settings
localhost abrtd: Executable '/usr/sbin/asterisk' doesn't
belong to any package
Mar 6 12:11:15 localhost abrtd: 'post-create' on
'/var/spool/abrt/ccpp-2013-03-06-12:11:09-26528' exited with 1
*Asterisk was running as root user
Any suggestions?
Regards,
Zohair Raza
a core dump because it is started with safe_asterisk
Thanks again
Regards,
Zohair Raza
On Thu, Mar 7, 2013 at 10:52 PM, Bharat Lalcheta
bharatlalch...@gmail.comwrote:
Did u test it without abrt?
On Mar 7, 2013 10:03 PM, Zohair Raza engineerzuhairr...@gmail.com
wrote:
Its Centos 6
Just to add that it was fixed by using this patch
https://issues.asterisk.org/jira/browse/ASTERISK-13145
It also made cisco softkeys working and call transfer/3 way conference as
well
Regards,
Zohair Raza
On Sun, Feb 17, 2013 at 2:58 AM, Vladimir Mikhelson v...@mikhelson.comwrote:
Zohair
because of packet loss in the network
Any suggestions?
Regards,
Zohair Raza
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Backtrace and logs attached here :
https://issues.asterisk.org/jira/browse/ASTERISK-21447
Regards,
Zohair Raza
On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry...@gmail.com wrote:
this is my secondary email
Regards
Zohair
On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry
.
Regards,
Bharat Lalcheta
On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza
engineerzuhairr...@gmail.com wrote:
Backtrace and logs attached here :
https://issues.asterisk.org/jira/browse/ASTERISK-21447
Regards,
Zohair Raza
On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry
thing I doubt is Insecure field, it is set to no at the moment. By name
it is for security only but setting it insecure=port may effect?
Hope it will solve your problem
Regards,
Bharat Lalcheta
On Tue, Apr 16, 2013 at 11:26 AM, Zohair Raza
engineerzuhairr...@gmail.com wrote:
Here is what
My experience was good, Nicolas was very helpful and quick
Regards,
Zohair Raza
On Tue, Jun 18, 2013 at 4:26 AM, Carlos Alvarez car...@televolve.comwrote:
No vacation notice, nothing, other than the system auto-replying saying
that the ticket will be closed because we didn't have any action
?
Regards,
Zohair Raza
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