[asterisk-users] How to count Parked calls?

or track the ParkedCalls() in the dialplan?? Through Asterisk CLI I can see the parked calls but I need to count the calls in dialplan. Muhamamd Faheem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

[asterisk-users] How to Play IVR and Read DTMF During Active Call?

voice problem. Please give your sugession. I am using asterisk 1.4 on making SIP calls in Local test environment with no NAT issues there. Thank you Muhammad Faheem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] how to implement CLONED LINE Feature in asterisk?

and will have the same telephone number as the original telephone line.   -  The Cloned Line is NOT a second telephone number.  The telephone number that is assigned to the second phone port on the device is the same telephone number as the number assigned to phone port one. Thanks! Faheem

Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?

is able to make calls, other user with earlier registration can not make call. My point here is in chain_sip.c what are variables or structure that need to maintain so that we can consider all registered users as active users. Thanks! Faheem --- On Wed, 8/5/09, D Tucny d...@tucny.com wrote: From

Re: [asterisk-users] Setting up Outgoing Trunk

insecure=no disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm promiscredir=yes t38_udptl=yes qualify=25000 nat=yes When u done that, reload sip. sip reload To verify it's correct: do these  in the asterisk CLI sip show peer user sip show registry Muhammad Faheem Software Engineer AXVoice Inc

Re: [asterisk-users] Multiple user registration ...

Parse the events If it is registration Event then store the Username/IP/Ports/Technology in Database # dial plan run agi script to get all strings eg. first Device:   SIP/u...@192.168.0.123:5061 second Device:  SIP/u...@10.0.0.150:6060 The complete script is attached. Muhammad Faheem

Re: [asterisk-users] Multiple user registration ...

customization, you need to modify it according to  your requirements. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com --- On Thu, 8/27/09, Mauro Sergio Ferreira Brasil mauro.bra...@tqi.com.br wrote: From: Mauro

Re: [asterisk-users] Multiple user registration ...

: Cloneline table:users(Username,IP1,Port1,Ip2,Port2) all varchars(30) Please adjust the table fields appropriately. Hope this code block will solve you problems. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com --- On Fri, 8

Re: [asterisk-users] Multiple user registration ...

}SIP/us...@${ip2}:${port2}) Hope every thing would be clear... Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com --- On Fri, 8/28/09, Mauro Sergio Ferreira Brasil mauro.bra...@tqi.com.br wrote: From: Mauro Sergio

[asterisk-users] How to Disable CDR for callfile?

callfiles, without changing the default behaviour of CDR logging. I know its NoCDR() function that will disable CDR() logging, But how it will be done in callfiles ? Thanks, M. Faheem ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] How to Disable CDR for callfile?

attemps? How I do it? CallFile: Channel: SIP/username CallerID: callback 100 MaxRetries: 3 RetryTime: 10 WaitTime: 40 Context: bridgecall Extension: 12129339037 Set:NoCDR Priority: 1 Account: 123; Thanks M. Faheem --- On Thu, 9/3/09, Danny Nicholas da...@debsinc.com wrote: From: Danny Nicholas

Re: [asterisk-users] Config Files

Muhammad Faheem --- On Wed, 10/14/09, Matt mhop...@gmail.com wrote: From: Matt mhop...@gmail.com Subject: [asterisk-users] Config Files To: asterisk-users@lists.digium.com Date: Wednesday, October 14, 2009, 7:39 PM Greetings, I have a fresh asterisk installation. When

Re: [asterisk-users] multiple call

Through Asterisk AMI, you can not dial multiple number at the same time. If you are going to implement a concurrent call scenario, then AMI would not be a valid choice. Multiple calls can be implemented with callfile. Faheem --- On Wed, 10/14/09, kaustuva...@bbsr.syscomes.com kaustuva

[asterisk-users] Conference Calling

this scenario. What are generic steps to do so! Thanks=Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] Is there any Diguim distributor in Lahore

Hey, is there any Diguim distributor in Lahore,Pakistan? I need to buy X100P.  Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] How to play Floating point numbers?

Hi all, Is there any way to play floating number using asterisk dialplan? Thanks,Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] How to play Floating point numbers?

Thanks Danny!  It solved my problem. Faheem --- On Mon, 6/7/10, Danny Nicholas da...@debsinc.com wrote: From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] How to play Floating point numbers? To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users

[asterisk-users] How to reuse mysql connection between AGI's

 there should be one connection, and it should be reused between each agi and when a call is over it should be disconnected. Is there any mechanism to reuse single MySQL connection between agi scripts?The agi scripts are written in Perl Thanks, Faheem, M

Re: [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?

Try make menu and select the speex module. make sure to do a  make clean also. Faheem, Muhammad VoIP Developer @ Vopium  --- On Fri, 8/6/10, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: From: Deepika Nijhawan deepika.nijha...@oxygen8.com Subject: [asterisk-users] [Asterisk-Users] How

Re: [asterisk-users] changing sip port

asterisk by default listen on port 5060.You simply need open the file /etc/asterisk/sip.conf and change these. udpbindaddr=0.0.0.0:6080tcpbindaddr=0.0.0.0:6080save the file and open asterisk console and execute sip reload. Muhammad Faheem --- On Fri, 11/12/10, Baha @ SH i...@saudihome.com

[asterisk-users] Get Channel Variables in AMI Event NewExten

',           'AppData' = '',           'Priority' = '4' -- Now my question is how can I get the variable fu_callerid in the AMI event block.? Please suggest any work around if possible. Thank you! Muhammad Faheem-- _ -- Bandwidth

Re: [asterisk-users] AMI help needed

You can use POE for geting AMI events. I'm sending you a simple poe.pl file in attachment, where you will get all raw events, and some callbacks are implemented for particular events.  For your case you can add few callback like conference join event, conference leave event. Muhammad Faheem

[Asterisk-Users] Asterisk Configuration

I have installed Redhat Linux 9 and Asterisk 1.2.1 on new computer. I need to know initial configuration of Asterisk i.e How to register a sip user?. What files do I have to edit? I am new about the Asterisk please help me Faheem Ahmed

Re: [asterisk-users] Get Channel Variables in AMI Event NewExten

Thanks! Matthew and Dan. On Thu, May 9, 2013 at 10:18 PM, Matthew Jordan mjor...@digium.com wrote: On 05/09/2013 08:16 AM, Dan Cropp wrote: I believe you will have to monitor for the Newexten event, then send an AMI Getvar command. It doesn’t make sense to pass all the possible channel

[asterisk-users] AMI Originate issue

Hi, I'm getting an issue while executing AMI Originate. I'm getting extension does not exists on Originate's Response, and on the other hand Asterisk CLI say fwrite() returned error: Broken pipe Please suggest me what is wrong. Muhammad Faheem ### my originate code block

[asterisk-users] Call Transfer question

? Thank you! Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate

Your both channels legs are identical strings. It should be like this. Action: Originate Channel: Local/outbound1@originateDialContext CallerID: 00311234567 Context: originateDialContext2 Exten: outbound1 Priority: 1 Variable:

Re: [asterisk-users] RTP from pcap file

You can take the pcap trace using tshark or tcpdump command line linux based tool and open the trace in wireshark. Wireshak is visual tool of tcpdum/tshark(corss platform) and you can listen audio of each call. On Fri, Jul 26, 2013 at 10:17 PM, Gianluca Merlo gianluca.me...@gmail.comwrote:

[asterisk-users] invalid From/Contact header values

you! Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users

Re: [asterisk-users] I want to store cdr into database

, Muhammad Faheem On Thu, Sep 17, 2015 at 3:21 PM, Amelye Chatila <amec...@gmail.com> wrote: > I have asterisk 13.5 configured with a simple dial plan, 3 SIP clients two > Laptops and smartphone with softphones installed. Now I am trying to store > cdr into a database but not able to ma

Re: [asterisk-users] AgentLogin() on the multiple servers?

Regards, Muhammad Faheem On Tue, Sep 15, 2015 at 3:46 AM, Shahid H <shah...@gmail.com> wrote: > Hello, > > Let say all the SIP devices will be registered on the proxy like kamailio. > > Agent is a member of Support and Billings Queues on the asterisk servers. >

Re: [asterisk-users] Want to detect sound

Try MixMonitor. Land the call to a local channel and answer it. This code will record the silence as well. exten => _X.,1,MixMonitor() exten => _X.,n,Dial(Local/100@context1) [context1] exten => _X.,1,Answer() exten => _X.,n,Dial(SIP/${EXTEN} On Tue, Jun 7, 2016 at 2:16 PM, Mamadou NGOM

Re: [asterisk-users] Delay after Answer

blem solved. Ported everything over to PJSip > and build RDNS records for the phones and the server, but I am still > experiencing the problem on incoming calls. > > > On 6/7/2016 1:00 PM, Faheem Muhammad wrote: > > I've faced the same issue. The issue was related to DNS, the re

Re: [asterisk-users] Delay after Answer

I've faced the same issue. The issue was related to DNS, the reverse lookup query failure caused the delay around(7-9 seconds). The purpose of reverse lookup is to block IP Spoofing attacks. Regards, Faheem On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson <br...@texascountrytitle.com> wrote

Re: [asterisk-users] Authentication header in BYE packets

is to add a SIP Proxy(opensips/kamillio) in between your Provider and Asterisk server and manipulate the BYE message with challenge. Regards, Muhammad Faheem On Thu, Jun 23, 2016 at 12:19 AM, Owais Ahmad <millennium@gmail.com> wrote: > Hi all, > > My provider proxy expects

Re: [asterisk-users] variable to get waittime of caller exiting queue

Israel, You can calculate the time diff by this dialplan snippet. --- exten = _X.,1,Set(callstarttime=${STRFTIME(${EPOCH},,%Y%m%d)}${STRFTIME(${EPOCH},,%H%M%S)}) exten => _X.,n,Queue(queue1) exten =

Re: [asterisk-users] Is MixMonitor command is blocking ?

MixMonitor() is non blocking command. It sets recording instructions and jumps to next priority instantly. On Tue, May 3, 2016 at 4:25 PM, Loic Chabert wrote: > Hello, > > I try to find informations concerning Mixmonitor command, but ... without > success. > MixMonitor

[asterisk-users] PJSIP_DIAL_CONTACTS issue

US}) exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1) exten => h,1,NoOp() exten => h,n,NoOP(${DIALSTATUS}) The endpoint may register from multiple device, so I always have to dial it all contacts. Did anyone

Re: [asterisk-users] PJSIP_DIAL_CONTACTS issue

Thanks Richord and Carlos. On Wednesday, 20 July 2016, Carlos Chavez <cur...@telecomabmex.com> wrote: > On 7/20/16 9:58 AM, Faheem Muhammad wrote: > > Hi, > I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. > > When I try to call an offline endpoint

Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

default). Similarly you can reduce the Call setup time by configuring 'T2' upto you choice as per you telephony network. Configure t1min, timert1 and timerb according to your network. Also set session-type=uas. Regards, Muhammad Faheem On Tue, Aug 9, 2016 at 12:03 PM, Jacek Konieczny <

Re: [asterisk-users] Asterisk 13 externip

On Wednesday, 14 September 2016, Madushan Geethanga wrote: > Hi, > > What is the equal option for externip in asterisk 13 with pjsip. I have > tried > > external_media_address=XX.XX.XX.XX > external_signaling_address=XX.XX.XX.XX > > but asterisk 13 writes local ip to the

[asterisk-users] codec negotiation or transcoding issue

Hi, I'm facing strange issue while establishing inbound calls from SIP trunks. Provider A is sending (G729,Alaw,uLaw) offer and asterisk dial the peer with its preferred codec order(G729,aLaw, uLaw). The peer's phone send the codec list as (uLaw, speex) in 200 OK replay. The Peer's phone has

Re: [asterisk-users] AMI Originate not working

ity, $application, $data, $timeout, $callerid, $vars, $account, $async, $actionid); echo "Status: $status"; } - Regards, Faheem On Thu, May 11, 2017 at 2:18 PM, Thomas <thomasit...@gmail.com> wrote: > Hello, > > I