[asterisk-users] How to append the recording file.

is appended to file created in above step. Now I know that 'a' option is used to append the recording to a file but I couldn't find any example on how to use it? Also if I use 'a' option and file doesn't exist then is it created or it is error? Any suggestions plea

Re: [asterisk-users] ${ANSWEREDTIME} returning null

Thanks, That worked. :) Anurag Rana http://newbie42.blogspot.in/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] ${ANSWEREDTIME} returning null

Oh, Sorry My mistake, I misspelled it in mail. It is already ${DIALEDPEERNUMBER}, still returning null. Anurag Rana http://newbie42.blogspot.in/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] ${ANSWEREDTIME} returning null

Call file syntax: Channel: SIP/ MaxRetries: 2 Context: demo1 Extension: s Priority: 1 WaitTime: 30 RetryTime: 60 in dialplan: exten=>h,n,NoOp(${DIALLEDPEERNUMBER) variable ${DIALLEDPEERNUMBER} is returning null. Suggestions please? Thanks Anurag Rana http://newbie42.blogspot

[asterisk-users] ${ANSWEREDTIME} returning null

Hi, I am initiating a call using call files. In 'h' extension I am trying to collect the value of ANSWEREDTIME variable but it is returning null. While It works fine when call is not generated using call files instead is generated from softphone. any idea what might be wrong? tha

Re: [asterisk-users] sip.conf and extension.conf configuration

The dots in extension will work as special characters. On 14/09/2014 8:06 pm, "rafa alfurqan" wrote: > Hi, > > i want to ask about sip.conf & extension.conf the configuration. > > is it possibility to make sip.conf configuration like this > [1510891531543...@wlan.mnc089.mcc510.3gppnetwork.org] >

[asterisk-users] is pattern matching inside macro valid?

xtension '5', but no rule 'i' or 'e' in context 'demo3' -- Executing [h@demo3:1] NoOp("SIP/101-000d", "terminating call") in new stack [Sep 9 02:11:14] NOTICE[9984]: pbx_spool.c:

Re: [asterisk-users] Pattern Extension not working in Dialplan

​Thanks. I will try it. Meanwhile I was trying below code. call goes to 'test' context and from there is passed to macro 'age'. In 'age' macro when I am using any patter to accept even single digit, its not working. So instead of using pattern I hardcoded the extension, but still when I am pressin

Re: [asterisk-users] Pattern Extension not working in Dialplan

@A J Stiles : If you could provide an example as you said, It would be very nice. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every T

Re: [asterisk-users] Pattern Extension not working in Dialplan

Thanks for the suggestion. @Stiles - Look like this may work. Will try this. Thanks. Anurag Rana http://newbie42.blogspot.in/ On Mon, Sep 8, 2014 at 1:42 PM, A J Stiles wrote: > On Sunday 07 Sep 2014, Anurag Rana wrote: > > Hi, > > > > I created a dummy dialplan wh

Re: [asterisk-users] Pattern Extension not working in Dialplan

​Thank you all for your suggestions. 1. [macro-age] is a macro and not an extension badly named. 2. I am able to use Read to fulfill the purpose but we can't use Read() after Background(). To use read we need Playback() [ am I right?]. But Playback do not provide barge-in facility i.e. user have

[asterisk-users] Pattern Extension not working in Dialplan

t digit and terminates even before considering second digit. Error message : WARNING[5726][C-000a]: pbx.c:6696 __ast_pbx_run: Invalid extension '8', but no rule 'i' or 'e' in context 'testmacro' Please sug

[asterisk-users] transfering call to dialplan without disconnecting.

t and dialplan structure (without disconnecting the call). How can this be achieved? Thanks Anurag Rana -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introdu

[asterisk-users] Recording sound.

my part of conversation is barely audible. I am recording using MixMonitor(). Is there anything that can be done to mitigate the problem? Anurag Rana http://newbie42.blogspot.in/ -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] How to execute an AGI script for each call.

receive call. exten => 111,1,Dial(SIP/111) exten => 112,1,Dial(SIP/112) exten => h,1,AGI(pt.py) ;; executes no matter what extension hang up ​Thanks​ -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the m

Re: [asterisk-users] Attack on Sip server.

Ok. Thanks. :) On Fri, Jun 27, 2014 at 11:05 PM, Mitul Limbani wrote: > No way out. Fix ur gateway which is masquerading out to in traffic. > > And do some research as others mentioned instead of expecting quick fix. > > Mitul > On 27-Jun-2014 10:45 PM, "Anurag Rana&

Re: [asterisk-users] Attack on Sip server.

Can't use anything which block IP addresses because my system is behind a gateway and attacker gets the address of that gateway. In this way I will end up blocking myself. Please suggest something else. On Fri, Jun 27, 2014 at 10:24 PM, Anurag Rana wrote: > Right Mitul. System is beh

Re: [asterisk-users] Attack on Sip server.

tration. If you do have some >> valid registration, you might want to add some counter to see how time IP >> need to fail or how many different users IP is trying to register on before >> blocking the IP. >> >> Jai Rangi >> www.didforslae.com >> >> >

Re: [asterisk-users] Attack on Sip server.

Both Rules* (typo in last mail) On Fri, Jun 27, 2014 at 8:19 PM, Anurag Rana wrote: > I added bot rules TCP as well as UDP. Still not working. > > How changing SIP listen port will prevent it. Please explain. > > I will try fail2band. > > > On Fri, Jun 27, 2014 at 8

Re: [asterisk-users] Attack on Sip server.

t; With regards > > N.Prakash > ------ > From: Anurag Rana > Sent: ‎27-‎06-‎2014 08:07 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [asterisk-users] Attack on Sip server. > > > Hi All. > > Someone is attacking o

[asterisk-users] Attack on Sip server.

t; --algo bm -j DROP ​Its something like this Registration from '"30" failed for '192.168.xxx.xxx:6373' - Wrong Password​ ​and there are approx 10 request per minute of this type. Please suggest some way to stop this.​ -- Anurag Rana http://newbie42.blogspot.in/ On t

Re: [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.

our-agi-program.pl) > exten => s,2,others... > > > Regards.. > rv > > > > > 2014-06-26 11:19 GMT-04:00 Anurag Rana : > >> Hi All, >> >> There is an option of starting the recording of call after the call is >> bridged. [ b option]. >> Is there

[asterisk-users] Executing an AGI python script in Asterisk after call is bridged.

Hi All, There is an option of starting the recording of call after the call is bridged. [ b option]. Is there any way of running an AGI script only if call is bridged otherwise not. Thanks -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towa

[asterisk-users] Changing recorded file storage directory.

Hi All, In asterisk, default directory to store the call-recording files is /var/spool/asterisk/monitor. Can we change this directory? How? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these material

Re: [asterisk-users] Echo Cancellation when calling from softphone to mobile.

Is there any Software solution? On Wed, Jun 25, 2014 at 11:38 PM, Mitul Limbani wrote: > Put line side echo cancelation chip on ur PRI card. > On 25-Jun-2014 10:35 PM, "Anurag Rana" wrote: > >> Hi, >> >> I am using Twinkle to call mobile phone but the

[asterisk-users] Echo Cancellation when calling from softphone to mobile.

Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http: