m->Originate ($channel,$exten, $context, $priority,
$application, $data, $timeout, $callerid, $vars, $account, $async,
$actionid);
echo "Status: $status";
}
-
Regards,
Faheem
On Thu, May 11, 2017 at 2:18 PM, Thomas wrote:
&g
Hi,
I'm facing strange issue while establishing inbound calls from SIP trunks.
Provider A is sending (G729,Alaw,uLaw) offer and asterisk dial the peer
with its preferred codec order(G729,aLaw, uLaw). The peer's phone send the
codec list as (uLaw, speex) in 200 OK replay. The Peer's phone has select
On Wednesday, 14 September 2016, Madushan Geethanga
wrote:
> Hi,
>
> What is the equal option for externip in asterisk 13 with pjsip. I have
> tried
>
> external_media_address=XX.XX.XX.XX
> external_signaling_address=XX.XX.XX.XX
>
> but asterisk 13 writes local ip to the from header. any suggesti
default). Similarly you can reduce the
Call setup time by configuring 'T2' upto you choice as per you telephony
network. Configure t1min, timert1 and timerb according to your network.
Also set session-type=uas.
Regards,
Muhammad Faheem
On Tue, Aug 9, 2016 at 12:03 PM, Jacek Koniec
Thanks Richord and Carlos.
On Wednesday, 20 July 2016, Carlos Chavez wrote:
> On 7/20/16 9:58 AM, Faheem Muhammad wrote:
>
> Hi,
> I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.
>
> When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the di
LSTATUS})
exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)
exten => h,1,NoOp()
exten => h,n,NoOP(${DIALSTATUS})
The endpoint may register from multiple device, so I always have to dial it
all contacts. Did a
is to add a SIP Proxy(opensips/kamillio) in between your
Provider and Asterisk server and manipulate the BYE message with challenge.
Regards,
Muhammad Faheem
On Thu, Jun 23, 2016 at 12:19 AM, Owais Ahmad
wrote:
> Hi all,
>
> My provider proxy expects authentication header on BYE p
to PJSip
> and build RDNS records for the phones and the server, but I am still
> experiencing the problem on incoming calls.
>
>
> On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
>
> I've faced the same issue. The issue was related to DNS, the reverse
> lookup query failur
I've faced the same issue. The issue was related to DNS, the reverse lookup
query failure caused the delay around(7-9 seconds). The purpose of reverse
lookup is to block IP Spoofing attacks.
Regards,
Faheem
On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson
wrote:
> I am having an issu
Try MixMonitor. Land the call to a local channel and answer it.
This code will record the silence as well.
exten => _X.,1,MixMonitor()
exten => _X.,n,Dial(Local/100@context1)
[context1]
exten => _X.,1,Answer()
exten => _X.,n,Dial(SIP/${EXTEN}
On Tue, Jun 7, 2016 at 2:16 PM, Mamadou NGOM wrote:
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
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
,
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
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 "
-
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
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
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
suggest if possible?
Thank you!
Muhammad Faheem
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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
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
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.
'Extension' => '111222',
'Application' => 'Answer',
'Uniqueid' => '1367903383.682',
'AppData' => '',
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
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
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
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,
Hi all, Is there any way to play floating number using asterisk dialplan?
Thanks,Faheem
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Hey, is there any Diguim distributor in Lahore,Pakistan? I need to buy X100P.
Muhammad Faheem
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scenario. What are generic steps to do so!
Thanks=Muhammad Faheem
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asterisk-users mailing list
To UNSUBSCRIBE or
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
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
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
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
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(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
---
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
---
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
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
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
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
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
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
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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
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asterisk-
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
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