Re: [asterisk-users] AMI Originate not working

2017-05-11 Thread Faheem Muhammad
m->Originate ($channel,$exten, $context, $priority, $application, $data, $timeout, $callerid, $vars, $account, $async, $actionid); echo "Status: $status"; } - Regards, Faheem On Thu, May 11, 2017 at 2:18 PM, Thomas wrote: &g

[asterisk-users] codec negotiation or transcoding issue

2017-03-14 Thread Faheem Muhammad
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 select

Re: [asterisk-users] Asterisk 13 externip

2016-09-15 Thread Faheem Muhammad
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 from header. any suggesti

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

2016-08-09 Thread Faheem Muhammad
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 Koniec

Re: [asterisk-users] PJSIP_DIAL_CONTACTS issue

2016-07-20 Thread Faheem Muhammad
Thanks Richord and Carlos. On Wednesday, 20 July 2016, Carlos Chavez 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 with PJSIP_DIAL_CONTACTS, the di

[asterisk-users] PJSIP_DIAL_CONTACTS issue

2016-07-20 Thread Faheem Muhammad
LSTATUS}) 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 a

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

2016-06-23 Thread Faheem Muhammad
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 wrote: > Hi all, > > My provider proxy expects authentication header on BYE p

Re: [asterisk-users] Delay after Answer

2016-06-07 Thread Faheem Muhammad
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 reverse > lookup query failur

Re: [asterisk-users] Delay after Answer

2016-06-07 Thread Faheem Muhammad
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 wrote: > I am having an issu

Re: [asterisk-users] Want to detect sound

2016-06-07 Thread Faheem Muhammad
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 wrote:

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

2016-05-18 Thread Faheem Muhammad
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 = _X.,n,Set(c

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

2016-05-03 Thread Faheem Muhammad
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 command take at last par

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

2015-09-17 Thread Faheem Muhammad
, Muhammad Faheem On Thu, Sep 17, 2015 at 3:21 PM, Amelye Chatila 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 make a connection of ODBC d

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

2015-09-15 Thread Faheem Muhammad
Regards, Muhammad Faheem On Tue, Sep 15, 2015 at 3:46 AM, Shahid H 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. > Support queue on "

[asterisk-users] invalid From/Contact header values

2013-12-11 Thread Muhammad Faheem
- uname -a Linux 6g-asterisk-devel 2.6.32-279.el6.x86_64 #1 SMP Fri Jun 22 12:19:21 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux asterisk -rx "core show version" Asterisk 11.5.1 built by root @ 6g-asterisk-devel on a x86_64 running Linux on 2013-10-07

Re: [asterisk-users] RTP from pcap file

2013-07-29 Thread Muhammad Faheem
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 wrote: > Hello James, > > Il giorno

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

2013-06-19 Thread Muhammad Faheem
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: recipient=0031612345678,callerid1=003

[asterisk-users] Call Transfer question

2013-05-16 Thread Muhammad Faheem
suggest if possible? 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-users] AMI Originate issue

2013-05-11 Thread Muhammad Faheem
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

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

2013-05-10 Thread Muhammad Faheem
Thanks! Matthew and Dan. On Thu, May 9, 2013 at 10:18 PM, Matthew Jordan 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 variables

Re: [asterisk-users] AMI help needed

2013-05-07 Thread Faheem
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.

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

2013-05-06 Thread Faheem
        'Extension' => '111222',           'Application' => 'Answer',           'Uniqueid' => '1367903383.682',           'AppData' => '',

Re: [asterisk-users] changing sip port

2010-11-11 Thread Faheem
asterisk by default listen on port 5060.You simply need open the file /etc/asterisk/sip.conf and change these.>> udpbindaddr=0.0.0.0:6080>>tcpbindaddr=0.0.0.0:6080save the file and open asterisk console and execute "sip reload". Muhammad Faheem --- On Fri, 11/12/10, B

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

2010-08-08 Thread Faheem
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 wrote: From: Deepika Nijhawan Subject: [asterisk-users] [Asterisk-Users] How do I install speex for asterisk

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

2010-08-05 Thread Faheem
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, Fahe

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

2010-06-07 Thread Faheem
Thanks Danny!  It solved my problem. Faheem --- On Mon, 6/7/10, Danny Nicholas wrote: From: Danny Nicholas Subject: Re: [asterisk-users] How to play Floating point numbers? To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Date: Monday, June 7, 2010,

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

2010-06-07 Thread Faheem
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

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

2010-03-26 Thread Faheem
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] Conference Calling

2010-02-27 Thread Faheem
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 or

Re: [asterisk-users] Config Files

2009-10-14 Thread Faheem
at=yes Muhammad Faheem --- On Wed, 10/14/09, Matt wrote: From: Matt 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 I install I get all

Re: [asterisk-users] multiple call

2009-10-14 Thread Faheem
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 wrote

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

2009-09-03 Thread Faheem
call 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 wrote: From: Danny Nicholas

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

2009-09-03 Thread Faheem
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] Multiple user registration ...

2009-08-30 Thread Faheem
(Fetch fetchid ${resultid} ip1 port1 ip2 port2) exten=> 112233,n,Dial(SIP/us...@${ip1}:${port1}&SIP/us...@${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 ---

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

2009-08-28 Thread Faheem
uld be. DB: 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 ---

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

2009-08-27 Thread Faheem
ly with my 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 wrote: From: Mauro Sergio Ferre

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

2009-08-27 Thread Faheem
AMI Events 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. Muham

Re: [asterisk-users] Setting up Outgoing Trunk

2009-08-11 Thread Faheem
yes 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" Muha

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

2009-08-05 Thread Faheem
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 wrote: From: D Tucny

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

2009-08-04 Thread Faheem
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

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

2009-07-17 Thread Faheem
ow I overcome that oneway 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://w

[asterisk-users] How to count Parked calls?

2009-07-14 Thread Faheem
nt 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-

[Asterisk-Users] Asterisk Configuration

2005-12-23 Thread Faheem Ahmed
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