Re: [asterisk-users] AMI Originate not working

[Faheem Muhammad]

Thomas,
this code block should work for your Originate case.
This code block will dial a local channel where actual leg 1 number is
dialed. On Answer of leg1, the leg2 is called.

-

require_once('phpagi-2.20/phpagi-asmanager.php');
$asm = new AGI_AsteriskManager('phpagi.conf');
$phone_no = '1416000';
$callerid = '1416001';
$leg1_exten = '1000';

if($asm->connect()){
$channel = "Local/".$leg1_exten ."@context_leg1";
$exten = "2000";
$context = "context_leg2";
$priority = 1;
$application = "";
$data = "";
$timeout = 3;
$callerid = $callerid;
$vars = "t_trunk=$t_trunk,campaign_name=$campaign_name,ivr_name=$ivr_name";
$account = "";
$async = 1;
$actionid = "";

$status = $asm->Originate ($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:

> Hello,
>
> I want to call an phone and if phone picked up I want to ring another
> phone.
> Or I want to connect to an running channel and then call another phone or
> move
> to an ConfBridge
>
> Iam using PHP
> $channel = 'IAX2/556-1696';
> or $channel = 'SIP/0019736363636@outbound.patton';
> $exten = '';
> $context = 'test_callout';
> $priority = '1';
>
>
> $parameters = array(
> 'Channel' => $channel,
> 'Exten' => $exten,
> 'Context' => $context,
> 'Priority' => $priority,
> );
> self::manager_com('Originate', $parameters);
>
>
> I get only this message, but no action or other information
>   == Manager 'vserver_webastmanager' logged on from 127.0.0.1
>   == Manager 'vserver_webastmanager' logged off from 127.0.0.1
>
>
> The AMI access in general should work, because I use it for another
> commands
> for example QueueAdd
>
> best regards
>
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[asterisk-users] codec negotiation or transcoding issue

[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 selected
only uLaw and speed in this case.

Ideally Asterisk should establish the call on uLaw codec, but Asterisk
establish the call with two codec for this call. For downstream RTP is
established with G729 and for upstream RTP is established with uLaw codec.
This behavior cause the one way audio for some phones like Eyebeam 1.5.9
but Phonerlite latest version allow it and there is no audio issue.

Is it normal SIP RFC 3261 behavior or there is something wrong with codec
negotiation or transcoding?

I'm using Asterisk 13.14.0 with realtime chan_pjsip compiled with bundled
pjproject on centos 6.8_x64. I have tested it with Asterisk 11.x with
chan_sip and it works fine.

Please advise me how can I setup the call based on late negotiation
mechanism?

Thank you!
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Re: [asterisk-users] Asterisk 13 externip

[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 suggestions?
>
> Best Regards,
> Madushan
>
>
>

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

[Faheem Muhammad]

Jacek,
This might be a bug or configuration issue, but you need to understand the
SIP Session Timers. With Session Timers you can control the round trip time
and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms 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 <jaj...@jajcus.net> wrote:

> Hi,
>
> We have been migrating our PBX system from Asterisk 1.8 and chan_sip to
> Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have
> stumbled on a behaviour difference I don't like.
>
> With chan_pjsip when a phone went unexpectedly offline (Ethernet cable
> disconnected) Asterisk would detect this quickly (through the 'qualify'
> pings), mark the phone as 'Unavailable' and fail immediately with
> 'CHANUNAVAIL' when dialling this phone.
>
> With Asterisk 13 and chan_pjsip qualify still works for determining
> current phone availability (endpoint shown as 'Unavailable' shortly
> after disconnecting the cable), but the phone is being dialled like
> nothing is wrong – Asterisk sends the INVITE and waits for the response,
> until SIP timeout (a bit more than 30s total). That is much longer time
> until 'CHANUNAVAIL' than I expect. It is also longer than the dial
> timeout in some cases, so I would get 'NOANSWER' instead of
> 'CHANUNAVAIL' which breaks my dialplan logic.
>
> Is that that the expected behaviour, a bug or a configuration problem?
> Am I supposed to check for device availability in my dialplan?
>
> Greets,
> Jacek
>
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Re: [asterisk-users] PJSIP_DIAL_CONTACTS issue

[Faheem Muhammad]

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 with PJSIP_DIAL_CONTACTS, the dial
> command breaks and the call control go to hangup block instead of next
> priority. The error in CLI says "*Dial requires an argument
> (technology/resource)*".
> This error seems legit as there are no contacts for an offline endpoint.
> The dialplan should jump to the next priority.
>
> exten => 1001,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
> exten => 1001,2,,NoOP(${DIALSTATUS})
> exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)
>
> exten => h,1,NoOp()
> exten => h,n,NoOP(${DIALSTATUS})
>
> ---
> If i try to dial the same offline endpoint with the below code snippet, it
> jumps to next prirorty.
> exten => 1001,1,Dial(PJSIP/${EXTEN})
> exten => 1001,2,,NoOP(${DIALSTATUS})
> exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)
>
> exten => h,1,NoOp()
> exten => h,n,NoOP(${DIALSTATUS})
>
> The endpoint may register from multiple device, so I always have to dial
> it all contacts. Did anyone else face such problem?
>
> My solution to this problem was to use a gotoif and check if
> PJSIP_DIAL_CONTACTS has any contacts before trying to dial, if it does not
> then I skip the dial and goto the next step.  So:
>
> exten => 1001,1,GotoIf($["${PJSIP_DIAL_CONTACTS(${EXTEN})}" = ""]?nocon)
> exten => 1001,n,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
> exten => 1001,n(nocon),SomethingElse
>
> --
>
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> +52 (55)9116-91161
>
>

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[asterisk-users] PJSIP_DIAL_CONTACTS issue

[Faheem Muhammad]

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 dial
command breaks and the call control go to hangup block instead of next
priority. The error in CLI says "*Dial requires an argument
(technology/resource)*".
This error seems legit as there are no contacts for an offline endpoint.
The dialplan should jump to the next priority.

exten => 1001,1,Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})
exten => 1001,2,,NoOP(${DIALSTATUS})
exten => 1001,3,Dial(PJSIP/mytrunk/sip:${mob}@10.0.0.1)

exten => h,1,NoOp()
exten => h,n,NoOP(${DIALSTATUS})

---
If i try to dial the same offline endpoint with the below code snippet, it
jumps to next prirorty.
exten => 1001,1,Dial(PJSIP/${EXTEN})
exten => 1001,2,,NoOP(${DIALSTATUS})
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 else face such problem?

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

[Faheem Muhammad]

Strange, A BYE should be replied with 200 OK, 481 (non matching dialogid),
408 request time out or similar responses, but it should never be
challenged. Only INVITE, REGISTER and  PUBLISH requests are challenged with
401/407.
As per rfc3261 it should not challenge the BYE Requests.
*The workaround 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 authentication header on BYE packets as well. Is
> there a way in asterisk to add this header on BYE packets?
>
> When proxy replies with a 401 on BYE, asterisk just retransmits the BYE
> packet.
>
> Regards,
> Owais
>
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Re: [asterisk-users] Delay after Answer

[Faheem Muhammad]

Are you sure *nslookup  *command is returning as expected?
Also check the output of the below command.
>> hostname && hostname -s && hostname -f


On Tue, Jun 7, 2016 at 11:54 PM, Brent Davidson <br...@texascountrytitle.com
> wrote:

> Well, I thought I had the problem 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 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:
>
>> I am having an issue with a couple of phones where they ring, but there
>> is a long delay after the phone is picked up before the audio starts.
>>
>> My setup:
>>
>>- Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
>>- Server is CentOS 7
>>- Quad core CPU with 16GB Ram
>>- 2 Snom 300 phones.
>>- NO NAT.  Server and phone are on the same subnet with only a
>>gigabit switch between them.
>>- Digium TDM400 analog card with 2 incoming analog PSTN lines
>>
>> When a call comes in, the system answers, IVR plays, caller dials an
>> extension, Snom 300 rings, handset picked up.  Caller continues to hear
>> ringing for another 7 to 10 seconds.  Answerer hears a click, a quick burst
>> of audio, then silence, then another click and audio is engaged.
>>
>> I have tried both SIP and RTP debugging and there are absolutely no
>> messages indicating any timeout or retransmit.  I am at a total loss.  In
>> the past I've always been able to find an answer to issues like this on my
>> own, but this time I just don't know.  I was even beginning to suspect the
>> network switch might be bad, but pinging between the server and the phones
>> shows no packet loss and 0.969ms average response time.
>>
>> What am I missing*?*
>> Thanks,
>> Brent Davidson
>>
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Re: [asterisk-users] Delay after Answer

[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 issue with a couple of phones where they ring, but there is
> a long delay after the phone is picked up before the audio starts.
>
> My setup:
>
>- Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
>- Server is CentOS 7
>- Quad core CPU with 16GB Ram
>- 2 Snom 300 phones.
>- NO NAT.  Server and phone are on the same subnet with only a gigabit
>switch between them.
>- Digium TDM400 analog card with 2 incoming analog PSTN lines
>
> When a call comes in, the system answers, IVR plays, caller dials an
> extension, Snom 300 rings, handset picked up.  Caller continues to hear
> ringing for another 7 to 10 seconds.  Answerer hears a click, a quick burst
> of audio, then silence, then another click and audio is engaged.
>
> I have tried both SIP and RTP debugging and there are absolutely no
> messages indicating any timeout or retransmit.  I am at a total loss.  In
> the past I've always been able to find an answer to issues like this on my
> own, but this time I just don't know.  I was even beginning to suspect the
> network switch might be bad, but pinging between the server and the phones
> shows no packet loss and 0.969ms average response time.
>
> What am I missing*?*
> Thanks,
> Brent Davidson
>
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Re: [asterisk-users] Want to detect sound

[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:

> Hello everybody,
>
> I manage not to detect one silence with record () when I make as follows:
>
> Exten = > 0178900271, n, Record ($ ${ link_recorded_pseudos_clients }
> pseudo_ Client_Id} wav, 5,5) exten = > 0178900271, n, GotoIf ($ [" $ {STAT
> (e, RECORDED_FILE} " = "0"]? Erreur_enregistrement_PPX17_1)
>
> When I say nothing, it do not return to the stage
> "erreur_enregistrement_PPX17_1"
>
> If you can help me?
>
> Mamadou NGOM
>
> Ingénieur Télécommunications & Réseaux
>
> Mobile: *06-47-02-67-86*
>
> Skype: Mamadou Numericap
>
> NumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 –
> TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015.
> siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny
> 83000 Toulon. mail: fina...@numericap.com 
> Centre d’exploitation : « Résidence les Coquières » 11 avenue Joseph
> Fallen - 13400 Aubagne – Tel :04.42.73.88.52
>
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Re: [asterisk-users] variable to get waittime of caller exiting queue

[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(callendtime=${STRFTIME(${EPOCH},,%Y%m%d)}${STRFTIME(${EPOCH},,%H%M%S)})
exten =_X.,n,Set(diff=$[${calltime1} -${calltime}])
exten=_X.,n,NoOp(diff)
-

Regards,
Muhammad


On Wed, May 18, 2016 at 5:05 PM, Israel Gottlieb <isr...@gmail.com> wrote:

> Hi all
>
> Is there anyway i could get in the dialplan  the amount of time a caller
> waited in the queue before exiting?
>
> Thanks
>
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Re: [asterisk-users] Is MixMonitor command is blocking ?

[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 parameter "command". This command can be a
> shell script.
>
> When record is over, and this command executed, asterisk wait for a return
> code or asterisk move to the next dialplan instruction ?
> This command is a background task or use ressources in asterisk ?
>
> For exemple, i need to send this file by mail, asterisk have to wait the
> end of upload file, or can he go to the next instruction ?
>
> Thanks,
> Regards.
> --
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>
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Re: [asterisk-users] I want to store cdr into database

[Faheem Muhammad]

It is very simple, asterisk can log cdrs automatically by configuring
cdr_mysql.conf.
All you need to create a mysql table along with proper read/write
permissions. You can find the cdr table schema from the below link.

https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend

Regards,
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 make a connection of ODBC drivers to
> MySQL is there an option or anything. Thanks in advance
>
> My configuration::
> *sip.conf*
>
> [general]
> trasport=udp ;Data format | sample commennt
>
> [template01](!)
> type=friend
> context=from-internal
> host=dynamic
> disallow=all
> allow=ulaw
> context=from-internal
> secret=unsecurepassword
>
> [6001](template01)
>
> [7001](template01)
> bindport=6050
>
>
> *extensions.conf*
>
> [from-internal]
> exten => 7001,1,Dial(SIP/7001,30)
> exten => 6001,1,Dial(SIP/6001,30)
>
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Re: [asterisk-users] AgentLogin() on the multiple servers?

[Faheem Muhammad]

You can achieve this by choosing one of asterisk server for pins collection
on extension 1234. When any member/extension dial that extension you need
to call a script that will make AMI connection on all servers and do
AgentLogin/QueueAdd Request.
You need to do ami login and call the AMI request QueueAdd on all server
where you have define different queues. It will make the agent login on all
Queue servers.
Below is snippet for making QueueAdd request from AMI.

-

Action: QueueAdd
Queue: supportqueue
Interface: sip/1122
Penalty: 1


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.
> Support queue on "Server  A" and Billings Queue on "Server B" for example.
> This will be done via RealTime Queue.
>
> I want Agent to dial 1234 on a sip device and it will prompt to enter a
> pin number to Login via AgentLogin(). Agent will stay on the line after
> logged in and wait for the calls.. I understand how this work from single
> asterisk server.
>
> But how is it possible for Agent to stay on the line from multiple
> asterisk servers or how it should be done? If agent dial 1245 for logging
> in - does kamailio randomly need to pink any server and then prompt for Pin
> via AgentLogin()?
>
> Thanks
>
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[asterisk-users] Asterisk mixmonitor with 16khz

[Muhammad Yousuf]

Hi I have a transcription software which requires media files with wide
band frequencies. Is it possible asterisk can record calls with 16khz not
8khz ?



Best regards


Muhammad Yousuf
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Re: [asterisk-users] Multiple IAX2 Trunks Load balancing

[Muhammad Usman]

yeah -- searching how to perform this magic ...


On Fri, Dec 13, 2013 at 2:29 PM, Steven Howes steve-li...@geekinter.netwrote:

 On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote:
  Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to
 load balance incoming calls over IAX2 trunks. If any trunk goes down the
 calls traffic will be shared with other available trunks. When it gets Up
 the script is supposed to perform as desired i.e in load balance mode.

 Sounds wonderful.

 S
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Re: [asterisk-users] Multiple IAX2 Trunks Load balancing

[Muhammad Usman]

Friends let me define the scenario please;
Scenario:
2 asterisk servers (A  B) are connected using 05 IAX2 trunks between them.
The machine A is running asterisk  Openvpn server in TUN mode (5 instances
with difference IP addresses for clients). The machine B is running
asterisk with 05 OpenVPN clients using 05 bandwidths. The IAX trunks are
established between each pair of P-2-P ip address of machine A (The OPENVPN
Server)  machine B (The Openvpn client).
Requirement:
Required dial plan configuration at machine A for incoming calls from VoIP
Switch/VOS which can forward the calls to IAX2 trunks in round robin
fashion like Load Balancing. If any trunk goes down it starts forwarding
the traffic to other available trunks  when it gets UP the dialplan should
perform as desired. Like L.B  Fail-over scenarios.


On Fri, Dec 13, 2013 at 8:52 PM, Hans Witvliet aster...@a-domani.nl wrote:

 On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote:
  On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
   Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want
   to load balance incoming calls over IAX2 trunks. If any trunk goes
   down the calls traffic will be shared with other available trunks.
   When it gets Up the script is supposed to perform as desired i.e in
   load balance mode.
 
   Thanks in advance.
  
 
  Hans said:

 
  Perhaps it is possible to do the L.B. at the O.S. or network level, and
 let
  all trunks appear to asterisk to one single trunk.
 
  Don asks:
 
  What's the value of load balancing multiple IAX trunks between the same
  system pair? What resources are being balanced?
 
 ++

 Perhaps the O.P. can explain about his intentions...

 In some situations it makes sense though:
 If you have to connect two servers, and use different kind of
 infrastructure / multiple providers...

 hw


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[asterisk-users] Multiple IAX2 Trunks Load balancing

[Muhammad Usman]

Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to
load balance incoming calls over IAX2 trunks. If any trunk goes down the
calls traffic will be shared with other available trunks. When it gets Up
the script is supposed to perform as desired i.e in load balance mode.


Thanks in advance.
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[asterisk-users] invalid From/Contact header values

[Muhammad Faheem]

Hi,
I'm observing wrong From/Contact header values. When I try to set
CallerID(num) it has no effect in the From and Contact Headers, and these
values are the same as the dialed number.
SIP Peers are defined using asterisk realtime. If I define the SIP Peers
using sip.conf then From/Contact header value are correct.

extentions.conf
[test]
exten= 1000, 1,NoOp()
same= n,Set(CALLERID(num)=)
same= n,Set(CALLERID(name)=)
same= n,Dial(SIP/1000)

exten= 2000, 1,NoOp()
same= n,Set(CALLERID(num)=)
same= n,Set(CALLERID(name)=)
same= n,Dial(SIP/2000)


Here is the sip trace...
--- Executing [2000@test:1] NoOp(SIP/1000-0014, ) in
new stack
-- Executing [2000@test:2] Set(SIP/1000-0014,
CALLERID(num)=) in new stack
-- Executing [2000@test:3] Set(SIP/1000-0014,
CALLERID(name)=) in new stack
-- Executing [2000@test:4] Dial(SIP/1000-0014, SIP/2000) in new
stack
  == Using SIP RTP CoS mark 5
Audio is at 16264
Adding codec 14 (alaw) to SDP
Adding codec 13 (ulaw) to SDP
Adding codec 12 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.7.218:5060:
INVITE sip:2000@10.10.7.218:5060 SIP/2.0
Via: SIP/2.0/UDP my-ip:5060;branch=z9hG4bK73e9c721
Max-Forwards: 70
From:  sip:2...@sipdev.mydomain.com;tag=as2a72da29
To: sip:2000@10.10.7.218:5060
Contact: sip:2000@my-ip:5060
Call-ID: 1f75fe937c6194227e6b5a5c29f41...@sipdev.mydomain.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.5.1
Date: Wed, 11 Dec 2013 16:23:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 604923607 604923607 IN IP4 my-ip
s=Asterisk PBX 11.5.1
c=IN IP4 my-ip
t=0 0
m=audio 16264 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
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 10:50:45 UTC

Please suggest me, either I put the issue in issue tracker or there is some
workaround.

Thank you!
Muhammad Faheem
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Re: [asterisk-users] RTP from pcap file

[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
gianluca.me...@gmail.comwrote:

 Hello James,

 Il giorno 26/lug/2013 15:50, James Bensley jwbens...@gmail.com ha
 scritto:

 
  Howdy all,
 
  Does anyone know of a niffty CLI tool for Linux that can take a PCAP
  file that was created on a SIP PBX for example, and then dump the
  payload of the various RTP streams in there into seperate files so I
  can listen to them?
 
  I can go this graphically with Wireshark, but I'd like to script it
  for automation.
 
  Cheers,
  James.

 I personally use rtpbreak

 http://dallachiesa.com/code/rtpbreak/doc/rtpbreak_en.html

 For similar tasks

 Gianluca

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

[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=00311234567

Timeout: 1

** **

[originateDialContext]

exten = outbound1,1,Wait(1)

exten = outbound1,n,Set(recipient=${recipient})

exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)

[originateDialContext2]

exten = outbound1,1,Wait(1)

exten = outbound1,n,Dial(SIP/${callerid1}@originateChannel)



On Wed, Jun 19, 2013 at 11:20 AM, Grant Bagdasarian g...@cm.nl wrote:

 Hello,

 ** **

 I’d like to use the AMI interface to originate a call to a context in a
 dialplan, and handoff the dial control to the context.

 ** **

 Whenever I execute the below action, the recipient does ring, but when I
 answer it dials the recipient again. I believe this is because once
 answered the system is going to execute the Context/Exten/Prio in the
 Originate action?

 ** **

 Action: Originate

 Channel: Local/outbound1@originateDialContext

 CallerID: 00311234567

 Context: originateDialContext

 Exten: outbound1

 Priority: 1

 Variable: recipient=0031612345678

 Timeout: 1

 ** **

 [originateDialContext]

 exten = outbound1,1,Wait(1)

 exten = outbound1,n,Set(recipient=${recipient})

 exten = outbound1,n,Dial(SIP/${recipient}@originateChannel)

 ** **

 Anyone have an idea how to fix this?

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[asterisk-users] Issue in transcoding

[Muhammad Yousuf]

I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm
gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have
g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1
and call leg from gsm gateway is using codec gsm. I am having one way audio
and getting below mentioned warning. Asterisk version is 1.8.11.0

[Jun  2 17:08:28] WARNING[21652]: translate.c:162 framein: no samples for
g723tolin
[Jun  2 17:08:28] WARNING[21652]: translate.c:162 framein: no samples for
g723tolin
[Jun  2 17:08:28] WARNING[21652]: translate.c:162 framein: no samples for
g723tolin
[Jun  2 17:08:28] WARNING[21652]: translate.c:162 framein: no samples for

plz help what could be the issue.
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[asterisk-users] Call Transfer question

[Muhammad Faheem]

Hi,
is possible that two sip extensions: user-1 and user-2 are connected and I
want to transfer the call from user-1 to a third user user-3.
I know it is possible through feature keys mapping in features.conf, but I
want to do this through AMI or Asterisk CLI Commands?

Please suggest if possible?

Thank you!
Muhammad Faheem
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[asterisk-users] AMI Originate issue

[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 Faheem

### my originate code block ...
---
# ami-script.pl
my $astman = Asterisk::AMI-new(PeerAddr = '127.0.0.1', PeerPort =
'5038', Username = 'faheem', Secret = 'secret');
die Unable to connect to asterisk unless
($astman);
my $resp_code = $astman-send_action({Action =
'Originate',
Channel =
'Local/11223344',
Context = 'users',
Exten = 100,
Priority =1 });
sleep(2);
my $response = $astman-get_response($resp_code);
print $response-{'Response'} .\n;
print $response-{'Message'} .\n;
$astman-disconnect ();

Script Output...
*Error*
*Extension does not exist*
--
;extensions.conf
;;; Asterisk Dialplan
[default]
exten = 11223344,1,NoOp(welcome)
exten = 11223344,n,Answer()
exten = h,1,NoOp(hangup...)

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

[Muhammad Faheem]

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 variables along
  with a Newexten event.  There may be a ton of extra variables that
  someone may not want or need on the AMI.  Better to have them ask for
  specific variables that are not standard.
 
 
 
  Action: Getvar
 
  ActionID: ValueYouCanIdentify
 
  Channel: IAX2/X.X.X.X:4572-5011
 
  Variable: fu_callerid
 
 
 
  This will result in a response from AMI…
 
 
 
  Response: Success
 
  ActionID: ValueYouCanIdentify
 
  Variable: fu_callerid
 
  Value: 141688xyxzz
 
 
 
  The ActionID is very important if you want to watch for an exact
 response to your request.
 

 If you know the names of the channel variables, you can also configure
 manager to send them with every channel event.

 From manager.conf:

 ;
 ; Display certain channel variables every time a channel-oriented
 ; event is emitted:
 ;
 ;channelvars = var1,var2,var3

 So if you want fu_callerid, set:

 channelvars = fu_callerid

 And, once that variable is set, you should get a NewExten event, you
 should see the following key/value pair:

 ChanVariable(SIP/1234-0001): fu_callerid=foobar


 --
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 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org



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[asterisk-users] Jitter Buffer in asterisk 1.8.11.0

[Muhammad Yousuf]

I am using asterisk as SIP/GSM  gateway. I have 2 gsm cards installed in
server. I am having some issue in audio quality. I want to enable jitter
buffer on asterisk but don't know, how to do. Any one can help me.
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[asterisk-users] click2call with AMI?

[Muhammad]

Hi,
I have a PHP code with AMI to using in click2call system.

here is my code:
$user= usernamr;
$secret  = secret;
$channel = 'SIP/' . $sip;

 $context = from-internal;

$waitTime = 20;
$timeout = 2;

$priority = 1;

$maxRetry = 2;

$pos  = strpos($number, local);
if ($number == null):
exit();
endif;
if ($pos === false) :
$errno= 0;
$errstr   = 0;
$callerId = $number;
$oSocket  = fsockopen(localhost, 5038, $errno, $errstr, 20);
if (!$oSocket)
{
echo $errstr ($errno)br\n;
} else
{
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Events: on\r\n);
fputs($oSocket, Username: $user\r\n);
fputs($oSocket, Secret: $secret\r\n\r\n);
fputs($oSocket, Action: originate\r\n);
fputs($oSocket, Channel: $channel\r\n);
fputs($oSocket, WaitTime: $waitTime\r\n);
fputs($oSocket, CallerId: $callerId\r\n);
fputs($oSocket, Exten: $number\r\n);
fputs($oSocket, Context: $context\r\n);
fputs($oSocket, Priority: $priority\r\n\r\n);
fputs($oSocket, Async: yes\r\n\r\n);
fputs($oSocket, Action: Logoff\r\n\r\n);
sleep(2);
fclose($oSocket);
}
else :

endif;

most of the time it tell me NO ANSWER when click2call number, but in normal
way it's ok and ringing phone and ANSWER.
my code is ok or not? and what is best way to click2call.
I think .call file is ok to making click2call system.
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[asterisk-users] Dial failed due to trunk reporting BUSY - giving up

[Muhammad]

Hi
this message give me when I calling a number than actually not busy:
Dial failed due to trunk reporting BUSY - giving up

max channel is unlimited and sometimes it dial number ok but most of the
time it gives me this error.

Please inform me how can solve this problem.

thanks
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Re: [asterisk-users] Dial failed due to trunk reporting BUSY - giving up

[Muhammad]

*
*
*Version, hardware, configuration... we can't read your mind. *
Sorry, I used asterisk 1.8.11.0 on elastix 2.3 with default configuration
gsm modem: DINSTAR 8 port

there is 5 SIM card in gsm modem and defined one outbound trunk.



On Sun, Feb 17, 2013 at 12:35 AM, Christopher Harrington ch...@acsdi.comwrote:



 On Saturday, February 16, 2013, Muhammad wrote:

 Hi
 this message give me when I calling a number than actually not busy:
 Dial failed due to trunk reporting BUSY - giving up

 max channel is unlimited and sometimes it dial number ok but most of the
 time it gives me this error.

 Please inform me how can solve this problem.


 thanks


 Version, hardware, configuration... we can't read your mind.


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Re: [asterisk-users] Dial failed due to trunk reporting BUSY - giving up

[Muhammad]

(SIP/147-2fb7, Dial
failed for some
reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17) in new stack
-- Executing [s@macro-dialout-trunk:21] Goto(SIP/147-2fb7,
s-BUSY,1) in new st
ack
-- Goto (macro-dialout-trunk,s-BUSY,1)
-- Executing [s-BUSY@macro-dialout-trunk:1] NoOp(SIP/147-2fb7,
Dial failed dueto
trunk reporting BUSY - giving up) in new stack
-- Executing [s-BUSY@macro-dialout-trunk:2]
PlayTones(SIP/147-2fb7, busy) in
n   ew stack
-- Executing [s-BUSY@macro-dialout-trunk:3] Busy(SIP/147-2fb7,
20) in new stac   k
  == Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on
'SIP/147-2fb7   '
in macro 'dialout-trunk'
  == Spawn extension (from-internal, **xxx**, 6) exited non-zero on
'SIP/147-2fb7   '
-- Executing [h@from-internal:1] Macro(SIP/147-2fb7,
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/147-2fb7,
1?endmixmoncheck)
innew stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp(SIP/147-2fb7, End of
MIXMON check) in
new stack
-- Executing [s@macro-hangupcall:10] GotoIf(SIP/147-2fb7,
1?nomeetmemon) in
ne   w stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] NoOp(SIP/147-2fb7,
MEETME_RECORDINGFILE=)
in new stack
-- Executing [s@macro-hangupcall:16] GotoIf(SIP/147-2fb7,
1?noautomon) in
newstack
-- Goto (macro-hangupcall,s,18)
-- Executing [s@macro-hangupcall:18] NoOp(SIP/147-2fb7,
TOUCH_MONITOR_OUTPUT=)
in new stack
-- Executing [s@macro-hangupcall:19] GotoIf(SIP/147-2fb7,
1?noautomon2) in
newstack
-- Goto (macro-hangupcall,s,25)
-- Executing [s@macro-hangupcall:25] NoOp(SIP/147-2fb7,
MONITOR_FILENAME=)
innew stack
-- Executing [s@macro-hangupcall:26] GotoIf(SIP/147-2fb7,
1?skiprg) in new
sta   ck
-- Goto (macro-hangupcall,s,29)
-- Executing [s@macro-hangupcall:29] GotoIf(SIP/147-2fb7,
1?skipblkvm) in
newstack
-- Goto (macro-hangupcall,s,32)
-- Executing [s@macro-hangupcall:32] GotoIf(SIP/147-2fb7,
1?theend) in new
sta   ck
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] Hangup(SIP/147-2fb7, ) in
new stack
  == Spawn extension (macro-hangupcall, s, 34) exited non-zero on
'SIP/147-2fb7' in
ma   cro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/147-2fb7'

***
*
On Sun, Feb 17, 2013 at 12:35 AM, Christopher Harrington ch...@acsdi.comwrote:



 On Saturday, February 16, 2013, Muhammad wrote:

 Hi
 this message give me when I calling a number than actually not busy:
 Dial failed due to trunk reporting BUSY - giving up

 max channel is unlimited and sometimes it dial number ok but most of the
 time it gives me this error.

 Please inform me how can solve this problem.


 thanks


 Version, hardware, configuration... we can't read your mind.


 --
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 ACSDi Office: 763.559.5800
 Mobile Phone: 612.326.4248



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Re: [asterisk-users] target number is busy after some calls

[Muhammad]

yes
here is my CLI output:
Everyone is busy/congested at this time (1:1/0/0)
Executing [s@macro-dialout-trunk:20] NoOp(SIP/105-034b, Dial failed
for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17) in new stack
-- Executing [s@macro-dialout-trunk:21] Goto(SIP/105-034b,
s-BUSY,1) in new stack

I have one truck with 5 SIM defined.
I have to call more times to passed busy tone, actually target mobile
number is not busy.
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[asterisk-users] target number is busy after some calls

[Muhammad]

Hi,

I used Asterisk 1.8 and I have a gsm modem with 8 port.
When I called target number, gsm modem and asterisk show me one of these
ports active. after hangup, the actived port is going to idl status and
ready to use. but after some call from extension, when I want to call
another number, asterisk gives me Busy status, however all ports are idle
and ready to use.

I think asterisk have to flashed my extension. please let me know what is
your idea?
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Re: [asterisk-users] access control softphone registration through asterisk

[Muhammad]

This is exactly my scenario that you said in another word.

*check their permissions and if they are allowed to call*
how check permission with AMI?
Im not familier with dialplan. it's better to me to use AMI programming


On Sat, Feb 9, 2013 at 2:27 PM, Yves A. yves...@gmx.de wrote:

  Hi,

 are you using realtime extensions or the classic config-file
 extension.conf ?
 One way to go yould be to implement the allowed / not allowed logic in the
 context of your sip users.
 check their permissions and if they are allowed to call... continue with
 the dialplan, if not, route them
 to a voiceprompt saying that the call is prohibited due to whatever
 reasons...

 To do so, take a look at the dialplan functions if and db. Of course
 you somehow have to set a
 flag in asterisk, that decides about permissions... Don´t know which way
 you will programmatically
 set or clear this flag... there are hundreds of possibilites... the
 easiest way I think would be to use the
 asterisk build-in database (therefore the hint to the function db...)

 regards,
 yves

 Am 08.02.2013 22:18, schrieb Muhammad:

 Hi,
 I wana control my SIP register from asterisk.
 I other hand, when users login into their softphone, dont access to call
 and when I give them access, they can call.

 I dont know it's right way to plan my scenario/?


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[asterisk-users] access control softphone registration through asterisk

[Muhammad]

Hi,
I wana control my SIP register from asterisk.
I other hand, when users login into their softphone, dont access to call
and when I give them access, they can call.

I dont know it's right way to plan my scenario/?
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[asterisk-users] get CDR log after disposition status answer

[Muhammad]

Hi,
Im planning to store CDR log to another database, so I used these Dialplan
code first:
exten = _X.,n,DeadAGI(xml-rpc_sendCDRLog.php)

or

exten = h,n,AGI(xml-rpc_sendCDRLog.php)

Actually it seems both of them work, But not in ANSWER disposition
status.(when call answred, AGI not run)

Please let me know how can I run my AGI code when user answer the call and
then hangup it.
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[asterisk-users] problem to socket programming in AGI

[Muhammad]

Hi,
I know maybe this question is not related to asterisk, but I want to make
XML RPC web service to other http server.
I have elastix system. it is https and problem is source not destination
server. In xml rpc we have fsockopen connection to connect destination
server(xml rpc server). It return me connect error(0).

what is the problem. is this related to elastix(asterisk) server?
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[asterisk-users] #!/usr/bin/php -q unknown command

[Muhammad]

Hi,

I used elastix with asterisk 1.8
when I run my AGI code, cli give me theses errors:

SIP/147-0098AGI Tx  agi_callingpres: 0
SIP/147-0098AGI Tx  agi_callingani2: 0
SIP/147-0098AGI Tx  agi_callington: 0
SIP/147-0098AGI Tx  agi_callingtns: 0
SIP/147-0098AGI Tx  agi_dnid: unknown
SIP/147-0098AGI Tx  agi_rdnis: unknown
SIP/147-0098AGI Tx  agi_context: from-internal
SIP/147-0098AGI Tx  agi_extension: 90
SIP/147-0098AGI Tx  agi_priority: 2
SIP/147-0098AGI Tx  agi_enhanced: 0.0
SIP/147-0098AGI Tx  agi_accountcode:
SIP/147-0098AGI Tx  agi_threadid: -1226703984
SIP/147-0098AGI Tx  agi_arg_1:
SIP/147-0098AGI Tx 
SIP/147-0098AGI Rx  Usage: php [options] [-f] file [--] [args...]
SIP/147-0098AGI Tx  510 Invalid or unknown command
SIP/147-0098AGI Rx php [options] -r code [--] [args...]
SIP/147-0098AGI Tx  510 Invalid or unknown command
SIP/147-0098AGI Rx php [options] [-B begin_code] -R
code [-E end_code] [--] [args...]

.
.
.
.
-- SIP/147-0098AGI Script myAGI.php completed, returning 0


I run my php script in bash linux, it seems it is not work with -q parameter

what is problem?
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Re: [asterisk-users] #!/usr/bin/php -q unknown command

[Muhammad]

*which php*

/usr/bin/php

the path is ok,
php -h gives me some parameter except -q


On Wed, Jan 30, 2013 at 10:20 AM, Zyumbilev, Peter
pe...@aboutsupport.comwrote:

 from ssh(console) run which php
 it should give you path where it is installed.

 Peter

 On 30/01/2013 08:28, Muhammad wrote:
  Hi,
 
  I used elastix with asterisk 1.8
  when I run my AGI code, cli give me theses errors:
 
  SIP/147-0098AGI Tx  agi_callingpres: 0
  SIP/147-0098AGI Tx  agi_callingani2: 0
  SIP/147-0098AGI Tx  agi_callington: 0
  SIP/147-0098AGI Tx  agi_callingtns: 0
  SIP/147-0098AGI Tx  agi_dnid: unknown
  SIP/147-0098AGI Tx  agi_rdnis: unknown
  SIP/147-0098AGI Tx  agi_context: from-internal
  SIP/147-0098AGI Tx  agi_extension: 90
  SIP/147-0098AGI Tx  agi_priority: 2
  SIP/147-0098AGI Tx  agi_enhanced: 0.0
  SIP/147-0098AGI Tx  agi_accountcode:
  SIP/147-0098AGI Tx  agi_threadid: -1226703984
  SIP/147-0098AGI Tx  agi_arg_1:
  SIP/147-0098AGI Tx 
  SIP/147-0098AGI Rx  Usage: php [options] [-f] file [--]
 [args...]
  SIP/147-0098AGI Tx  510 Invalid or unknown command
  SIP/147-0098AGI Rx php [options] -r code [--] [args...]
  SIP/147-0098AGI Tx  510 Invalid or unknown command
  SIP/147-0098AGI Rx php [options] [-B begin_code] -R
  code [-E end_code] [--] [args...]
  
  .
  .
  .
  .
  -- SIP/147-0098AGI Script myAGI.php completed, returning 0
 
 
  I run my php script in bash linux, it seems it is not work with -q
 parameter
 
  what is problem?
 
 
 
 
 
 
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Re: [asterisk-users] AGI command

[Muhammad]

Thanks Salman.
Here is my cli debug when run agi:


-- Launched AGI Script /var/lib/asterisk/agi-bin/testAGI.php
-- SIP/101-0007AGI Script testAGI.php completed, returning 0


and here is my php agi source code:

#!/usr/bin/php -q
?php

error_reporting(E_ALL);
ob_implicit_flush(false);
set_time_limit(6);
$stdin = fopen('php://stdin', 'r');
$stdlog = fopen('tmp/agi_log.log', 'w');
$debug = false;
// Read input from Asterisk and output via $astOutput *
function astRead()
{
global $stdin, $debug, $stdlog;
$astOutput = str_replace(\n, , fgets($stdin, 4096));
 if ($debug) fputs($stdlog, read: $input\n);
return $astOutput;
}
// Write AGI command to Asterisk
function astWrite($agiCommand)
{
global $debug, $stdlog;
if ($debug) fputs($stdlog, write: $agiCommand\n);
 echo $agiCommand.\n;
}
// Handling execution input from Asterisk
while (!feof($stdin))
{
$temp = fgets($stdin);
$temp = str_replace(\n,,$temp);
$s = explode(:,$temp);
$agivar[$s[0]] = trim($s[1]);
if ($temp == )
{
break;
}
}
// Operational Code starts here

astWrite(STREAM FILE demo-congrats #);
astRead();
astWrite(SAY NUMBER 123456 #);
astRead();

// Finalization of AGI script and clean-ups
fclose ($stdin);
fclose ($stdlog);
exit(0);


as you see, it was return 0. agi is running successful or not?

[from-internal]
exten = _X.,1,NOOP(++Invoking AGI Script now(before agi): MGH)
exten = _X.,2,AGI(testAGI.php)
exten = _X.,3,NOOP(***After AGI)
exten = _X.,100,DeadAGI(getCDRLog.php)





On Sat, Jan 19, 2013 at 11:49 AM, Salman Zafar msalman...@gmail.com wrote:

 You seriously need to go through Asterisk text books available online or
 at-least read the manual.

 NoOP is just a No Operation, so its just printing text in your case.

 To execute an AGI you need to call it from dialpan like exten =
 _X.,n,AGI(name_of_AGI), agi should be appropriate directory, which I reckon
 you already know.



 On Sat, Jan 19, 2013 at 1:06 PM, Muhammad mohammad.ghaz...@gmail.comwrote:

 Thanks Steve.

 but explain me more about your dialplan codes. Im a beginner.


 exten = _X.,1,NOOP(Invoking AGI Script now)

 what is Invoking AGI Script now and what should I do in this method?

 what is _X. it seems it is dynamic extension number.
 and n?

 Thanks

 On Wed, Jan 16, 2013 at 11:12 AM, SamyGo govoi...@gmail.com wrote:

 Hi,
 Please see my comments in line.

 Regards,
 Sammy


 On Wed, Jan 16, 2013 at 12:13 PM, Zohair Raza 
 engineerzuhairr...@gmail.com wrote:


 On Wed, Jan 16, 2013 at 11:01 AM, Muhammad 
 mohammad.ghaz...@gmail.comwrote:

 *Thanks Zohair!
 I wrote some php code to working with AGI, but it dosen't work.
 *

 *I don't know how can run it. please explain me when I put my php code
 inside /var/lib/asterisk/agi-bin  so, what should I do after that. *


 Make sure Asterisk has access to your AGI script, and make it
 executable (chmod u+x agi.php). Also make sure it has shebang
 (!#/usr/bin/php)


 Besides  that you'll need to create SIP users and define their *
 context=my-agi* and in your context call this AGI.

 [my-agi]
 exten = _X.,1,NOOP(Invoking AGI Script now)
 same = n,AGI(my-agi-filename.php)
 same = n,NOOP(Any other post AGI things here)
 same = n,Hangup()



 *and the second one, how can limit users to call just my number in
 list at database and permit to call another numbers.*
 *
 *

 That depends on logic in your script, you can also separate users by
 contexts


 Alternative to the above approach there are other possibilities, like
 applying a GotoIF() condition in the *[my-agi]*  dialplan code above
 restricting a particluar number,  or Instead of *_X. *in the above code
 implement some regex to enable only local area dialing i.e

 exten = _78XX,1,NOOP(Invoking AGI Script now for dialled string
 starting with 78 and 8 digit in length)

 If you don't like any of the above because you've a list of numbers
 which should be dialled only in a DB.table then in your php-AGI script do
 some restrictions based on that table.



 * *
 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


 On Tue, Jan 15, 2013 at 1:05 PM, Muhammad mohammad.ghaz...@gmail.com
  wrote:

 Hi,

 in CLI, I type agi show or other agi commad, but response me
 command not found.
 How can see agi is work normally in my server?


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Re: [asterisk-users] AGI command

[Muhammad]

*Hi,
sorry to delay.**

here is my answers.
 1) What does the Asterisk console log look like when you try to execute
the AGI?
**first let me know how can I execute the AGI?**

 2) If you enter '/usr/bin/php -v', what do you get? I get:
**$ php -v give me:
PHP 5.1.6 (cli) (built: Nov 29 2010 16:47:37)
Copyright (c) 1997-2006 The PHP Group
Zend Engine v2.1.0, Copyright (c) 1998-2006 Zend Technologies**

**but not in path /usr/bin/php -v**

3) If you enter the following command in a shell, what do you get?

sudo -u asterisk /var/lib/asterisk/agi-bin/testAGI.php

**Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use
'asterisk -r' to connect.**



*
On Wed, Jan 16, 2013 at 8:33 PM, Steve Edwards asterisk@sedwards.comwrote:

 On Wed, 16 Jan 2013, Muhammad wrote:

  **When you say 'doesn't work' do you mean 'doesn't do what I want' or
 'does not execute?'

 I mean I do all steps in Mr. Nir presentation documents and not works.


 Your PHP script executes correctly on my dev box, but I would change the
 log file path to something absolute like '/tmp/agi_log.log' so we know
 where it is.

 1) What does the Asterisk console log look like when you try to execute
 the AGI?

 2) If you enter '/usr/bin/php -v', what do you get? I get:

 PHP 5.1.6 (cli) (built: Feb 22 2012 19:34:21)
 Copyright (c) 1997-2006 The PHP Group
 Zend Engine v2.1.0, Copyright (c) 1998-2006 Zend Technologies

 3) If you enter the following command in a shell, what do you get?

 sudo -u asterisk /var/lib/asterisk/agi-bin/**testAGI.php
 /dev/null

 (If you run Asterisk as another user, use that user name in the command
 above.)

 I get:

 PHP Notice:  Undefined offset:  1 in /var/lib/asterisk/agi-bin/**testAGI.php
 on line 31
 STREAM FILE demo-congrats #
 SAY NUMBER 123456 #

 The 'notice' is because we passed an empty AGI environment.


 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: 
 +1-760-468-3867https://2.177.142.31/_AMI/tidy.php?exten=147number=%2B17604683867PST
 Newline  Fax: 
 +1-760-731-3000https://2.177.142.31/_AMI/tidy.php?exten=147number=%2B17607313000

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Re: [asterisk-users] AGI command

[Muhammad]

Thanks Steve.

but explain me more about your dialplan codes. Im a beginner.


exten = _X.,1,NOOP(Invoking AGI Script now)

what is Invoking AGI Script now and what should I do in this method?

what is _X. it seems it is dynamic extension number.
and n?

Thanks

On Wed, Jan 16, 2013 at 11:12 AM, SamyGo govoi...@gmail.com wrote:

 Hi,
 Please see my comments in line.

 Regards,
 Sammy


 On Wed, Jan 16, 2013 at 12:13 PM, Zohair Raza 
 engineerzuhairr...@gmail.com wrote:


 On Wed, Jan 16, 2013 at 11:01 AM, Muhammad mohammad.ghaz...@gmail.comwrote:

 *Thanks Zohair!
 I wrote some php code to working with AGI, but it dosen't work.
 *

 *I don't know how can run it. please explain me when I put my php code
 inside /var/lib/asterisk/agi-bin  so, what should I do after that. *


 Make sure Asterisk has access to your AGI script, and make it executable
 (chmod u+x agi.php). Also make sure it has shebang (!#/usr/bin/php)


 Besides  that you'll need to create SIP users and define their *
 context=my-agi* and in your context call this AGI.

 [my-agi]
 exten = _X.,1,NOOP(Invoking AGI Script now)
 same = n,AGI(my-agi-filename.php)
 same = n,NOOP(Any other post AGI things here)
 same = n,Hangup()



 *and the second one, how can limit users to call just my number in list
 at database and permit to call another numbers.*
 *
 *

 That depends on logic in your script, you can also separate users by
 contexts


 Alternative to the above approach there are other possibilities, like
 applying a GotoIF() condition in the *[my-agi]*  dialplan code above
 restricting a particluar number,  or Instead of *_X. *in the above code
 implement some regex to enable only local area dialing i.e

 exten = _78XX,1,NOOP(Invoking AGI Script now for dialled string
 starting with 78 and 8 digit in length)

 If you don't like any of the above because you've a list of numbers which
 should be dialled only in a DB.table then in your php-AGI script do some
 restrictions based on that table.



 * *
 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


 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
 not found.
 How can see agi is work normally in my server?


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Re: [asterisk-users] AGI command

[Muhammad]

***When you say 'doesn't work' do you mean 'doesn't do what I want' or
'does not execute?'*

I mean I do all steps in Mr. Nir presentation documents and not works.

Here is my php code:

#!/usr/bin/php -q
?php
error_reporting(E_ALL);
ob_implicit_flush(false);
set_time_limit(6);
$stdin = fopen('php://stdin', 'r');
$stdlog = fopen('agi_log.log', 'w');
$debug = false;
/* Read input from Asterisk and output via $astOutput */
function astRead()
{
global $stdin, $debug, $stdlog;
$astOutput = str_replace(\n, , fgets($stdin, 4096));
if ($debug) fputs($stdlog, read: $input\n);
return $astOutput;
}
/* Write AGI command to Asterisk */
function astWrite($agiCommand)
{
global $debug, $stdlog;
if ($debug) fputs($stdlog, write: $agiCommand\n);

echo $agiCommand.\n;
}
/* Handling execution input from Asterisk */
while (!feof($stdin))
{
$temp = fgets($stdin);
$temp = str_replace(\n,,$temp);
$s = explode(:,$temp);
$agivar[$s[0]] = trim($s[1]);
if ($temp == )
{
break;
}
}
/* Operational Code starts here */

astWrite(STREAM FILE demo-congrats #);
astRead();
/* Say the number 123456 */
astWrite(SAY NUMBER 123456 #);
astRead();

/* Finalization of AGI script and clean-ups */
fclose ($stdin);
fclose ($stdlog);
exit(0);

and have wrote in extension_custom.conf:

[from-internal-custom]
include = testAGI

[testAGI]
exten = 147,1,Answer
exten = 147,2,AGI(testAGI.php)
exten = 147,3,Hangup


147 is my extension.
operators can call a number via their extension(each extension may use for
one or more users?)

I register extension 147 in my client softphone and call a number. what
happen after that?

On Wed, Jan 16, 2013 at 11:19 AM, Steve Edwards
asterisk@sedwards.comwrote:

 On Wed, 16 Jan 2013, Muhammad wrote:

  I wrote some php code to working with AGI, but it dosen't work.


 When you say 'doesn't work' do you mean 'doesn't do what I want' or 'does
 not execute?'

 If you enable AGI debugging, what does the Asterisk console log look like?

 Did you use an established PHP library or 'roll your own?'

 A good way to test an AGI is to create a text file containing all the
 cruft (the AGI 'environment') Asterisk sends to the AGI along with the
 expected responses. Then you can execute your AGI completely external from
 Asterisk with a shell command line like:

 /var/lib/asterisk/agi-bin/my-**firs-agi example environment

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: 
 +1-760-468-3867https://2.177.142.31/_AMI/tidy.php?exten=147number=%2B17604683867PST
 Newline  Fax: 
 +1-760-731-3000https://2.177.142.31/_AMI/tidy.php?exten=147number=%2B17607313000


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[asterisk-users] AGI command

[Muhammad]

Hi,

in CLI, I type agi show or other agi commad, but response me command not
found.
How can see agi is work normally in my server?
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Re: [asterisk-users] AGI command

[Muhammad]

*Thanks Zohair!
I wrote some php code to working with AGI, but it dosen't work.
I don't know how can run it. please explain me when I put my php code inside
/var/lib/asterisk/agi-bin  so, what should I do after that. and the second
one, how can limit users to call just my number in list at database and
permit to call another numbers.**
*
On Tue, Jan 15, 2013 at 12:39 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:

 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 not
 found.
 How can see agi is work normally in my server?


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Re: [asterisk-users] php programming for working with asterisk

[Muhammad]

*thanks to replay Sammy!
But excatly I don't know how can do it! connecting to DB via dialplan.   **
*
On Mon, Jan 14, 2013 at 11:08 AM, SamyGo govoi...@gmail.com wrote:

 Hi,

 If your caller is using softphone then you can create a simple dialplan
 which consults a DB and verifies that the dialled number is in the
 allowed-caller-list and if the result is OK just let the call dial through.
 Use Mixmonitor() application in your dialplan to record this call at some
 location(file-path) and on Hangup insert the call related data into DB.

 This all can be done from within the dialplan
 (extensions.conf/extensions.ael) OR via any AGI as well.

 AMI has different purposes and, for starters, should not be confused
 together and compared.

 Thanks
 Sammy



 On Mon, Jan 14, 2013 at 11:23 AM, Muhammad mohammad.ghaz...@gmail.comwrote:

 Hi,

 I write some php code in AMI to working with asterisk command. I don't
 know exactly what is the different between AMI and AGI and witch one is
 better for my planning.
 Im planning to call party users that their number is is my panel on web.
 We have some operator and they can call party users via client softphone by
 clicking on their number, so they have to limited to call just listed
 number and restrict to call another number. how can do permission to
 this plan? and how can get stored call record in asterisk (IVR recorded
 voice) via php programming (AGI is better or AMI).


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[asterisk-users] php programming for working with asterisk

[Muhammad]

Hi,

I write some php code in AMI to working with asterisk command. I don't know
exactly what is the different between AMI and AGI and witch one is better
for my planning.
Im planning to call party users that their number is is my panel on web.
We have some operator and they can call party users via client softphone by
clicking on their number, so they have to limited to call just listed
number and restrict to call another number. how can do permission to this
plan? and how can get stored call record in asterisk (IVR recorded voice)
via php programming (AGI is better or AMI).
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[asterisk-users] Unable to execute 'dahdi_scan /etc/asterisk/dahdi_scan.conf'

[p070075 Muhammad Atif Ramzan]

Hi

Can anyone help me with this error
Unable to execute 'dahdi_scan  /etc/asterisk/dahdi_scan.conf'

i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call
reached the destination but no voice is coming from destination my voice
reflects back


thanks
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Re: [asterisk-users] Realtime peers and trunks coming from the same IP

[p070075 Muhammad Atif Ramzan]

Hi Sammy go

Can you help me with my problem
I have asterisk 1.8 and i am using asterisk-gui 2.0, and in asterisk-gui
2.0 the voice prompt menu which is used for custom voice recording for IVR
is not working and not recording. Can u tell me how to defualt this feature.


thanks
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[asterisk-users] upgrading from asterisk 1.4 to 1.6

[p070075 Muhammad Atif Ramzan]

Hi
 I have installed asterisk 1.4 and asterisk-gui 2.0, the problem is that it
cannot upload the .gsm which i record through voice menu prompt, it gives
error uploading is supported in asterisk 1.6 or higher.
Can anyone help me?


thanks
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[asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)

[p070075 Muhammad Atif Ramzan]

Hi

I am new to asterisk 1.4 can someone tell about how to enable the video
conference in asterisk-gui 2.0.
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Re: [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)

[p070075 Muhammad Atif Ramzan]

Actually i want to know that how i configure the asterisk for video
confernce
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[asterisk-users] How to add prefix in Extensions.Conf

[Muhammad Ali]

Hello,


I have a DID number 5672531308 , I want to add 92 prefix in it as been told
by my provider , so I can I do this in extensions.conf?
-- 
Regards,

Muhammad Ali
DIDx SUPPORT
http://www.didx.net
Skype: didxnet
Phone: +1-212-655-5763 / +1-850-433-8555
Direct : +1-567-2531308
http://www.youtube.com/watch?v=acCDzP-7oqYfeature=player_embedded
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Re: [asterisk-users] (no subject)

[Muhammad Usman]

you running GSM FWTs with asterisk ?

On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem abid_aster...@hotmail.comwrote:

  HI,

 I am trying to setup a Class 4 termination setup using a kind of channel
 hunting scenerio. I have some SIP DID numbers assigned from the local
 telecom provider for termination. MY call comes from my wholesale client and
 lands on a switch, then it is routed to asterisk. I want asterisk to route
 this call to my local DID provider on the next available channel with DID
 number as the new Caller ID. This is just like GSM gateway that recieves the
 call and then re-originates the call using the next available SIM card
 number.

 Can someone help me how can I configure Asterisk to perform this?

 Thanks

 Abid.

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Regards:
(Muhammad υѕмαη )
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Re: [asterisk-users] Nat=yes

[Muhammad Ali]

Hi,

When NAT = YES, Asterisk server will extract IP from the network layer. 
 
When Nat = No, the Asterisk server will respond to the IP in the SIP header. Am 
I right?

May be such type of options can be helpful for SIP application developers.

Can't think of a scenario but If it is set to be YES for all peers, what will 
happen is that the response to all the SIP request will be routed to the IP in 
the network layer. IP's in the SIP header will be ignored,  should not create a 
problem.
 

Regards

--- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote:

From: Steve Totaro stot...@asteriskhelpdesk.com
Subject: Re: [asterisk-users] Nat=yes
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, April 24, 2011, 2:13 PM



On Thu, Apr 21, 2011 at 5:42 AM, Alexandru Oniciuc 
alexandru.onic...@trivenet.it wrote:

Dear * users, in your opinion, when using a * as a public server, is good 
practice enabling nat=yes in sip.conf for all the peers?
Can anyone imagine a scenario when enabling this parameter (even for peers that 
don’t require it) can cause problems? 
Regards and thanks in advance,Alex 

I asked this same exact question several years ago.  There are many replies 
with different takes.  I would skim through Alex's posts, there is really 
nothing worth reading except it will break the SIP RFC handed down by the 
internets themselves.


I use nat=yes all the time and it works just fine.

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg213941.html


Nobody actually answered the question about the bad side, they just argued 
about the SIP RFC.

Many others agreed to make it default behavior and that setting nat=yes gives a 
an extra degree of security.


RFCs are great and all, but in the real world, phones just need to work.

Thanks,
Steve Totaro
 


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Re: [asterisk-users] Nat=yes

[Muhammad Ali]

Hi,

I am unsure of what you are saying.

Just for discussion, if one has a control on the insertion of  the IP address 
in the SIP header, then nat options working can be verified  observed.
 
In the OSI reference model, the Network is layer 3, IP.
Call it Network, layer 3, or IP, it is the same.

All-right, by IP from the network layer  I meant, the IP address in the IP 
Header/Network layer/layer 3.

  IP from SIP I meant,  SIP request generator's IP address in the SIP Header. 
I missed the word address.

My customers don't really care for things that don't work. 

May be its useful for SIP application developers rather then end customers.

Have a good time.
Regards

--- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote:

From: Steve Totaro stot...@asteriskhelpdesk.com
Subject: Re: [asterisk-users] Nat=yes
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, April 24, 2011, 3:28 PM



On Sun, Apr 24, 2011 at 5:55 AM, Muhammad Ali ali_...@yahoo.com wrote:

Hi,

When NAT = YES, Asterisk server will extract IP from the network layer. 
 
When Nat = No, the Asterisk server will respond to the IP in the SIP header. Am 
I right?


May be such type of options can be helpful for SIP application developers.

Can't think of a scenario but If it is set to be YES for all peers, what will 
happen is that the response to all the SIP request will be routed to the IP in 
the network layer. IP's in the SIP header will be ignored,  should not create a 
problem.

 

Regards

--- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote:



I am unsure of what you are saying.  

All I know is that setting nat=yes has never failed me when nat=no has and we 
are talking countless phones and installs.  

In the OSI reference model, the Network is layer 3, IP.


Call it Network, layer 3, or IP, it is the same.

nat=yes breaks the RFC due to NAT but it gets people talking.  My customers 
don't really care for things that don't work. 

Thanks,
Steve Totaro


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[asterisk-users] Help Required---Problem in Installation without dahdi

[Muhammad Ali]

Hi,

Installation of dahdi requires kernel source that is not available with my 
remote virtual machine. Therefore I installed Asterisk without installing dahdi 
but when I start Asterisk it crashes while loading chan_agent.so (noload is 
also not useful in this case).

Any suggestions or hints to overcome this issue?

Regards 
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Re: [asterisk-users] app_calendar and SSL

[Muhammad Nuzaihan]

Try to disable certificate verification on the app. I had never tried
it personally  but check for that option.

Sent from my iPhone

On Jan 17, 2011, at 5:51 PM, --[ UxBoD ]-- ux...@splatnix.net wrote:

 Hi,

 Over the weekend tried to setup a test using the new app_calendar code but 
 receiving the following error:

 [Jan 17 09:23:35] WARNING[27663]: res_calendar_icalendar.c:146 
 fetch_icalendar: Unable to retrieve iCalendar 'testcal' from 
 'https://office.test.net/home/teamsh...@test.net/Calendar/': Server 
 certificate verification failed: issuer is not trusted

 The target server is using a self signed cert so where would one store the 
 PEM on the Asterisk server for the calendar app to find it ?
 --
 Thanks, Phil

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[asterisk-users] Fix Fake Answer Supervision In asterisk1.6

[Muhammad Usman]

Hi,
I have installed asterisk1.6+DAHDI for TDM2400P Digium card. When call hits
the box, the gets answered even the other end phone in not picked. How can I
fix this as ideally it should answer the call when other end phone is
picked.
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[asterisk-users] digim tdm2400p fxo fake answer supervision problem.

[Muhammad Usman]

Hi. I am using Digium TDM2400PFXO with Asterisk1.6. When call sets in the
box , it answers the call even the phone is not picked. ideally it should
answer the call when the phone is picked up. Its charging the clients.
Please let me know how can I cover this ? Thanks in advance.
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Re: [asterisk-users] Attack problem

[Muhammad Nuzaihan Kamalluddin]

netstat -anp |grep 6667

Best Regards,
Muhammad Nuzaihan Kamal
Network Consultant
Mobile: +65 97473874

Asfa Systems Pte Ltd
91, Alps Avenue. #03-10. Singapore 498787

Tel:  +65 62538211
Fax: +65 62504814
www.asfasystems.com.sg

pub   4096R/36630777 2010-07-10
  Key fingerprint = 670A 4D60 0A2D 43A1 2FE0  DFDA D3A9 3F32 3663 0777
uid  Muhammad Nuzaihan Kamalluddin (Asfa Systems Pte. Ltd.) 
muham...@asfasystems.com
sub   4096R/97E5CBBD 2010-07-10



On 20-Dec-2010, at 5:40 PM, Khaled W. Chehab wrote:

 Ircd  is not installed and cant be located in all system ,any one know or
 have an idea how do they infect my system,
 Any bug in asterisknow?
 How to find the script that initiates this invites ?
 135.307281 192.168.138.56 - 218.75.79.17 TCP 36578  ircd [ACK] Seq=36
 Ack=111 Win=5840 Len=0
 135.307434 192.168.138.56 - 218.75.79.17 TCP 36578  ircd [FIN, ACK] Seq=36
 Ack=111 Win=5840 Len=0
 135.309188 218.75.79.17 - 192.168.138.56 TCP ircd  36578 [FIN, ACK]
 Seq=111 Ack=1 Win=4096 Len=0
 135.309211 192.168.138.56 - 218.75.79.17 TCP 36578  ircd [ACK] Seq=37
 Ack=112 Win=5840 Len=0
 135.334037 192.168.138.56 - 192.168.5.2  DNS Standard query A
 irc3.mysteryaddict.com
 135.334496  192.168.5.2 - 192.168.138.56 DNS Standard query response A
 87.229.45.226
 135.334657 192.168.138.56 - 87.229.45.226 TCP 53718  ircd [SYN] Seq=0
 Win=5840 Len=0 MSS=1460 TSV=1532274 TSER=0 WS=7
 135.342359 218.75.79.17 - 192.168.138.56 TCP ircd  42802 [SYN, ACK] Seq=0
 Ack=1 Win=1460 Len=0 MSS=1380
 135.342399 192.168.138.56 - 218.75.79.17 TCP 42802  ircd [ACK] Seq=1 Ack=1
 Win=5840 Len=0
 135.342554 192.168.138.56 - 218.75.79.17 IRC Request
 
 Regards
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
 Sent: Friday, December 17, 2010 6:20 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Attack problem
 
 On Friday 17 Dec 2010, Khaled W. Chehab wrote:
 HI,
 
 My system been attacked from someone I guess, kindly check the link 
 below
 
 How can I stop the ircd attack
 
 # /etc/init.d/ircd stop
 # chmod -x  /etc/init.d/ircd
 
 Should do the business  :)
 
 --
 AJS
 
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   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 *
 No employee or agent is authorized to conclude any binding agreement on 
 behalf of Xplorium with another party by e-mail without express written 
 confirmation by an officer of Xplorium. Any views expressed by an individual 
 in this electronic message do not necessarily reflect views of Xplorium or 
 its subsidiaries and associates.
 
 This electronic message and its attachments are solely addressed to the 
 addressee(s), and contain confidential information protected from disclosure 
 belonging to Xplorium.
 
 If you are not the intended addressee of this electronic message and its 
 attachments, kindly delete it immediately from your system and notify the 
 sender by electronic mail. You must not copy this message or attachment or 
 disclose its content to any other person.
 
 Xplorium does not guarantee the integrity of this electronic message and any 
 of its attachments, or that they are free from computer viruses or other 
 defects.
 *
 
 
 
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Re: [asterisk-users] MYSQL ADDON INSTALLATION ERROR

[Muhammad Nuzaihan Kamalluddin]

Hi Ridwan,

You would need to install mysql-devel via yum.

Best Regards,
Muhammad Nuzaihan Kamal
Network Consultant
Mobile: +65 97473874

Asfa Systems Pte Ltd
91, Alps Avenue. #03-10. Singapore 498787

Tel:  +65 62538211
Fax: +65 62504814
www.asfasystems.com.sg

pub   4096R/36630777 2010-07-10
  Key fingerprint = 670A 4D60 0A2D 43A1 2FE0  DFDA D3A9 3F32 3663 0777
uid  Muhammad Nuzaihan Kamalluddin (Asfa Systems Pte. Ltd.) 
muham...@asfasystems.com
sub   4096R/97E5CBBD 2010-07-10



On 06-Oct-2010, at 6:35 PM, Rizwan Hisham wrote:

 Hi All,
 Please refresh my memory. I am trying to install asterisk after 2 years. I 
 hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 
 1.8.0-rc2 on centos 5.5 but getting the following errors.
 
 app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory
 app_mysql.c: In function ‘mysql_ds_destroy’:
 app_mysql.c:135: warning: implicit declaration of function ‘mysql_close’
 app_mysql.c:138: warning: implicit declaration of function ‘mysql_free_result’
 app_mysql.c: In function ‘aMYSQL_connect’:
 app_mysql.c:319: error: ‘MYSQL’ undeclared (first use in this function)
 app_mysql.c:319: error: (Each undeclared identifier is reported only once
 app_mysql.c:319: error: for each function it appears in.)
 app_mysql.c:319: error: ‘mysql’ undeclared (first use in this function)
 
 I think i have seen these errors before and did manage to get rid of them but 
 I cant remember how i did it and even dont remember the reason for these 
 errors. Looks like a header file for mysql addon is missing which is actually 
 missing (i have checked). How am I suppose to find it?
 
 Plz help.
 
 -- 
 Best Regards
 Rizwan Qureshi
 
 
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[asterisk-users] OpenSer and MYSQL Lookup Queries!

[Muhammad Zulqarnain]

Hi!

Can OpenSer perform some database lookup queries based on dialed number like we 
can do with Asterisk. Asterisk Can do it and there is MYSQL Function available 
which allow us to open connection and execute any query to get required results 
from database, Can we do same with OpenSer or OpenSIP etc.?


Thanks
Regards,
Muhammad Zulqarnain

 
 

















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[asterisk-users] A2Billing Callback Hangup after/about 20 sec!

[Muhammad Zulqarnain]

Hi!

I am posting a2billing issue here in asterisk list because some one might have 
faced same issue with a2billing callback. My Callback problem has been already 
posted on a2billing forum which I am facing on my system. Please have a look on 
this thread:

http://forum.asterisk2billing.org/viewtopic.php?t=3093

I am using CID-Callback, when you call the access number, it calls back and 
ask for the destination number, as you enter the destination number the call 
will hangup about 20 - 24 seconds, as you are entering the destination number.

I am using Asterisk 1.4, A2billing 1.3.3 on Centos 5.2 every thing is fine 
except callback hangup issue. by digging Google and forums for same issue, I 
got answered to change the carrier and i have tried with six different carrier, 
and it works very rare among one out of ten calls.

Can any one suggest how to fix it. I have emailed a2billing support also for 
resolving issue but not yet got any reply. 

Thanks
Regards
Zulqarnain




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Re: [asterisk-users] World Most Economical Predictive Dialer!

[Muhammad Zulqarnain]

Hi Tilghman!
nbsp;
gt; Clearly, you missed the point.nbsp; Since there is a FREE predictive 
dialer out
gt; there, and your product costs something, you are not the world's cheapest
gt; predictive dialer.nbsp; 

I respect your wording and the way you or other people think on the list about 
difference between cheapest and free predictive dialer. 
nbsp;
Surely there is no comparison between Cheapest and free. I am also not making 
comparison between Free Open Source Predictive Dialer such as VICIDIAL, 
Gnudialer with Commercial TeleRep Performance Optimizer Predictive Dialer.
nbsp;
English is not my first language but what I do believe that there is a way to 
compare things; It does make sense to me that World Most Cheapest Predictive 
Dialer should be compared with World Most Expensive Predictive Dialer rather 
then World Most Free Predictive Dialer. 
nbsp;
Still there are several cost factor involved by using free predictive dialer 
like; Installation, Maintenance and sometime buying commercial Support for 
free/Open Source Predictive Dialer which doesn't make it 100% free for end user 
in some circumstances.
nbsp;
gt; The only way you could possibly be cheaper than free is if you paid people 
to use your 
gt; product.
nbsp;
I don't believe that the developer of free predictive dialer pay people to use 
their dialer. If you know any one let me know, I would be happy to get paid.
nbsp;
gt; Not a particularly wise business plan, but then, what do I know?
nbsp;
Even if you believe that in order to make TeleRep Performance Optimizer 
Predictive Dialer to be the World Most Cheap Predictive Dialer I need to pay 
for it then shoot me an off the list email as again it will be a commercial 
discussion on the non commercialnbsp; mailing list. It could be the wise 
business plan which I can let you know!
nbsp;
In favor of objection raised by my friend Tilghman Let me update my wording and 
subject of this email to reflect it to be the World Most Economical Predictive 
Dialer if it make better sense.

Thanks
Regards,
nbsp;
Zulqarnain
email: [EMAIL PROTECTED]

nbsp;


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Re: [asterisk-users] World Cheapest Predictive Dialer!

[Muhammad Zulqarnain]

Hi!

First of all I apologize for posting this commercial email on asterisk-user 
list but it gave me some valuable information which were not notified by any 
one on asterisk-buzz list. 

Second I would like to clear the confusion created by my email about FREE DNC 
Scrubbing. 

Our Predictive Dialer have capability to filter list of number against the List 
of National Do Not Call Registry provided by the client itself. So they don't 
have to buy any additional software from market, like dnclistmanager, Scrub DNC 
etc. for scrubbing their list before dialing their leads. In this way let me 
know if even we from developing country like Pakistan are violating US or FTC 
laws.

Some one asked in the list that how It is world most cheapest predictive 
dialer. We don't pay to people for using our software. TeleRep Performance 
Optimizer is not a free predictive dialer but the most cheapest Predictive 
Dialer (just 0.014c per minute, as you use your own carrier to terminate call 
anywhere in the world, but not only to USA) as compare to other Hosted Dialer 
available in the market like Callfire and many others available.

However, discussion made by several friends including Steve, MATT, Dean are 
very valuable and I thanks all of you for taking your time on discussing this 
matter on list. 

I am looking forward further for your feedback on this matter if still it is 
out of law.

Thanks
Regards,

Zulqarnain


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[asterisk-users] Unexpected behaviour shown by meetme kick confno usernumber

[Muhammad Raza]

Hi, 
I have Asterisk 1.4.4 on my linux box. 
Whenever i try to kick a participant in conference say 59681446 using
following command
meetme kick 59681446 1
where 1 is the participant number, following are the actions that
asterisk takes
 

*   IVR You have been kicked from this conference is played.
*   Participant is taken out from that conference 59681446
*   Plus the same participant is taken to a different conference
with conf # h (always), as the partcipant's phone does not hang up.

 
So, when the following command is issued 
meetme kick h 1
the participant's phone finally hangs up.
 
I don't know why is this behaviour shown by the meetme module ? Could
anybody help me in this regard ?
 
Thanx, 
Raza
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[asterisk-users] E1 controller

[Muhammad Aslam ul Haq]

dear all
i have 8-port e1 controller, i am some confuse about e1 commands
that is
when and why we use *cahnnel-group* and *pri-group* e1 controller command

let me konw the above question,
i have further more questions related to this issue.
i shall be very thankfull to you

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With Best Regards
Muhammad Aslam ul Haq
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[asterisk-users] problem with queues

[Muhammad Zeeshan Latif]

Hi all

Does any one experience the scenario in which an agent sits behind any trunk
and serves the queue. Bcz I have tried and the queue acts very veared as
when it tries the agent extension across the trunk it starts music onhold
and when the agent is busy/not responding then it stops music on hold again
when it starts hunting the agents across a trunk again starts the MOH and
again stops the MOH when it can't find the agent online.

To make thing worse if the MOH is dynamic then the user hears different MOH
tones after a period of 1 or two seconds, and if u have enabled voice
prompts in the queue then the result is even more bad.


Best Regards
Mohammad Zeeshan Latif


smime.p7s
Description: S/MIME cryptographic signature
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[asterisk-users] FW: problem with queues

[Muhammad Zeeshan Latif]

Hi all

Does any one experience the scenario in which an agent sits behind any
trunk
and serves the queue. Bcz I have tried and the queue acts very veared as
when it tries the agent extension across the trunk it starts music
onhold
and when the agent is busy/not responding then it stops music on hold
again
when it starts hunting the agents across a trunk again starts the MOH
and
again stops the MOH when it can't find the agent online.

To make thing worse if the MOH is dynamic then the user hears different
MOH
tones after a period of 1 or two seconds, and if u have enabled voice
prompts in the queue then the result is even more bad.


Best Regards
Mohammad Zeeshan Latif
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[asterisk-users] Having trouble to receive fax from samsung sf3200

[Muhammad Zaka]

 down
Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier up
Jun 26 09:20:10 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier trained
Jun 26 09:20:12 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier down
Jun 26 09:20:12 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW 
Trainability test failed - longest run of zeros was 903

Jun 26 09:20:12 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW  FTT:
Jun 26 09:20:12 DEBUG[14882]: app_rxfax.c:70 span_message:  44
Jun 26 09:20:12 DEBUG[14882]: app_rxfax.c:70 span_message:
Jun 26 09:20:15 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier up
Jun 26 09:20:15 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier training failed
Jun 26 09:20:17 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier down
Jun 26 09:20:17 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier up
Jun 26 09:20:17 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier trained
Jun 26 09:20:19 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier down
Jun 26 09:20:19 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW 
Trainability test failed - longest run of zeros was 88

Jun 26 09:20:19 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW  FTT:
Jun 26 09:20:19 DEBUG[14882]: app_rxfax.c:70 span_message:  44
Jun 26 09:20:19 DEBUG[14882]: app_rxfax.c:70 span_message:
Jun 26 09:20:22 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier up
Jun 26 09:20:22 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier training failed
Jun 26 09:20:24 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier down
Jun 26 09:20:24 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier up
Jun 26 09:20:24 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier trained
Jun 26 09:20:26 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier down
Jun 26 09:20:26 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW 
Trainability test failed - longest run of zeros was 955

Jun 26 09:20:26 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW  FTT:
Jun 26 09:20:26 DEBUG[14882]: app_rxfax.c:70 span_message:  44
Jun 26 09:20:26 DEBUG[14882]: app_rxfax.c:70 span_message:
Jun 26 09:20:29 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier up
Jun 26 09:20:29 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier training failed
Jun 26 09:20:30 DEBUG[14882]: app_rxfax.c:70 span_message: FLOW Non-ECM 
carrier down

 -- Channel 0/1, span 1 got hangup
Jun 26 09:20:30 DEBUG[14882]: app_rxfax.c:322 rxfax_exec: Got hangup

Please can you help me what is wrong.

Thanks

Best Regards

Muhammad Zaka

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[Asterisk-Users] nortel meridian option 11c and asterisk te110p

[Muhammad Zeeshan Latif]

Hi sir



I am trying to interconnect meridian option 11c 2mb pri card ntbk50aa
with * pri card te110p.

But the problem that I am facing is that both card do not see each other
the te110p card does not come out of red alarm and same is the case with
meridian ntbk50aa.

Hence I can not expect d-channel negociation to take place.

Can u guide me some what about this as the card u used is a little
different than the one I am using in m1. I have tried both t1 and e1 but
same problem remains. I have tested both card so no chance of error
there , cabling checked many a times no error there.

When I connect some times * te110p card shows yellow/red/rec alarm.
And on ntbk50aa card some times red light turn off and yellow comes on
and after some times the yellow light turn off and red comes on
permanently.




Best Regards
Mohammad Zeeshan Latif
Sr. WAN Engineer
NETWORK DIRECTORATE

0092-51-90391020, 0092-321-5181157
 
 
 
 
 
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[Asterisk-Users] nortel meridian option 11c and asterisk

[Muhammad Zeeshan Latif]

Hello all





After all very pain taking steps I have not yet been able to
connect the te110p and meridian option 11c ntbk50aa cards together .





I have tried many different options which were also posted
by greg camp and Koen Van Impe but I have not
been able to 



bring the isdn pri up the problem is that e1
interface does not come up so d-channel negociation 



can not take place I have triedall sorts of option
that I know in span command but of no use .





now I am in contact with digium support and I have given
them some imp info that I have done up till now.



I have pasted some of the contents and imp mail on
this site 



http://pastebin.ca/66088



and



http://pastebin.ca/66089







and 





http://pastebin.ca/66090







regards

zeeshan










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[Asterisk-Users] nortel meridian option 11c and asterisk

[Muhammad Zeeshan Latif]

 Hello all


Does any one know that te110p card operate in which mode


* 2 Mbit/s transparent E1 (G.703)
* 2 Mbit/s frame-structured E1 (G.703/G.704)
* 2 Mbit/s ISDN-PRA (ETS 300 233, I.431)


Bcz I am using meridian option 11c card which I am in dought that it
supports only ISDN-PRA mode which is the third mode in the above
convention and te110p does not support this mode. Is there any way to
change that.

Plz tell me.



Regards
zeeshan 

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[Asterisk-Users] nortel meridian option 11c and asterisk te110p

[Muhammad Zeeshan Latif]

Hi Koen Van Impe





Thanks for the meridian config and asterisk. I will
defenitly try them



And let every one know.





Just a few words and correct me if I am wrong





There are two things 





1
E1 : the 32 channels once both the equipment
see each other and the ccs/hdb3 encoding/format is read the LED infront of
interface goes green and this makes the lower layer work.

2
ISDN PRI: once step one is complete
we can proceed to the signaling of ISDN PRI that is euro isdn or 

5ess or any .





I might be wrong



But the problem that I face is the first step the e1 never
comes up I have and the LED never goes green. I have checked the cable it work
s fine with other pri which interms confirms the card also.





But with the new config that u have given me I pray it works
bcz it is very critical for my organization as we are tired of paying 

Nortel bags and bags of money and with this idea of using
asterisk and interface it with the existing meridian system we see a hope
of expanding with very little investment.







Thanks and regards



mohammad










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RE: [Asterisk-Users] nortel meridian option 11c and asterisk te110p

[Muhammad Zeeshan Latif]

Hi there sir 



Thanks for ur
suggestion but the problem with us is that we are running the whole distributed
call center in three different cities of pakistan.



So we can not take risk on that behalf we
just want that our expansion need to be fulfilled by expanding throght asterisk
which far cheaper than existing Nortel.



So thanks any ways for ur suggestion



Regards

zeeshan











Another
approach you might take would be to keep your Meridian
phones but get rid of your PBX, utilising a Citel SIP Handset Gateway to
interface the phones to the Asterisk server. See http://www.citel.com
for more details.








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[Asterisk-Users] asterisk and nortel meredian option 11c

[Muhammad Zeeshan Latif]

Hi Koen Van Impe





Thanks for the meridian config and asterisk. I will
defenitly try them



And let every one know.





Just a few words and correct me if I am wrong





There are two things 





1
E1 : the 32 channels once both the
equipment see each other and the ccs/hdb3 encoding/format is read the LED
infront of interface goes green and this makes the lower layer work.

2
ISDN PRI: once step one is complete
we can proceed to the signaling of ISDN PRI that is euro isdn or 

5ess or any .





I might be wrong



But the problem that I face is the first step the e1 never
comes up I have and the LED never goes green. I have checked the cable it work
s fine with other pri which interms confirms the card also.





But with the new config that u have given me I pray it works
bcz it is very critical for my organization as we are tired of paying 

Nortel bags and bags of money and with this idea of using
asterisk and interface it with the existing meridian system we see a hope
of expanding with very little investment.







Thanks and regards



mohammad



---











Best Regards

Mohammad Zeeshan Latif

Sr. WAN Engineer

NETWORK DIRECTORATE



0092-51-90391020,0092-321-5181157
















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[Asterisk-Users] FW: asterisk and nortel meredian option 11c

[Muhammad Zeeshan Latif]

Hi 





I want to connect asterisk 1.0.9 ( kernel 2.6.8-2 debian
)with TE110P and Nortel meridian option 11c release 25.40 with NTBK50AA card
which is 

An E1 card. But the main problem is the first stage that no sync
occurs the * card never syncs with meridian card







I am using euroisdn, ccs , hdb3, crc4 , pri_net on asterisk



And I am assuming that meridian is using same as it is
connected to Nortel passport mvpe card which is an e1 isdn card and using the
same config as astresk but the card never see each other. On the contrary when
I connect the asterisk with the Nortel passport mvpe card it does detect the
mvpe card but the d chan flaps btwn up and down and the hell of HDLC BAD FCS
messages appears on the cli of * .



I have also tried yellow alarm on the span but not of any
help .





Can any one tell me the config of meridian option 11c and
asterisk and what I am doing wrong.





thanks






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[Asterisk-Users] RE: help required plzzzzzzzzzz

[Muhammad Zeeshan Latif]

Sir 



Thanks so much but I have done lots and
lots of googling around and I also had a grip on this file earlier.



I have already tried this but this is for
the T1 scenerio. 



I am looking for the ISDN PRI over E1 and
it is not doing any good to me.



The exact card on the Nortel 11c is
NTBK50AA. Which is an E1/PRI card.



It seems to me that u have taken help from
that file I have seen the mailing list archives and seems that people are using
5ess instead of euroisdn.



In my case the physical interface does not
go up and te110p indicates red led all the time some times goes yellow.



I have confirmed the cable it work on
other links.



Is underneath the only change u made to
the described config of meridian and what was the value u put in dch
under in ld 96.













ENL SERV dch











Best Regards

Mohammad Zeeshan Latif

Sr. WAN Engineer

NETWORK DIRECTORATE



0092-51-90391020,0092-321-5181157





















From: Greg Camp
[mailto:[EMAIL PROTECTED] 
Sent: Thursday, June 08, 2006 5:55
PM
To: Muhammad
 Zeeshan Latif
Subject: RE: help required
plzz





Good luck.





Thanks,
Greg















From: Muhammad Zeeshan Latif
[mailto:[EMAIL PROTECTED] 
Sent: Thursday, June 08, 2006 2:41
AM
To: [EMAIL PROTECTED]
Subject: help required
plzz





Hi sir





I need ur help as I read ur post to google group
asterisk-users. Which is as under









Update: SUCCESS!!! There were two subtle items
that allowed our Opt81C 
to talk PRI to * using a TE110P: 

1)
On the 81C in LD 96 we had to ENL SERV dch for the d-channel. 

2)
It appears that the TE110P needs a decent refresh time for the 
b-channels to come up cleanly. For example, on the Nortel if you 
disable the T1 or d-channel and re-enable it quickly (specifically, 
anything shorter than about 45 seconds) the TE110P won't come up clean. 
The Nortel will show all the b-channels as MBSY or FE MBSY. However, if 
you wait 45s - 1min then the d-channel and b-channels will come up clean 
every time. 

Many
thanks to all who offered suggestions and worked with us on this! 









I am trying to connect * 1.0.9 with a TE110P card to Nortel
option 11c 25.40 release and having very serious issues with both.





First and last of all I was never able to bring the channels
up on asterisk nor meridian option 11c.



Can u plzz mail me the
configuration of meridian and asterisk I will be very greatful for that. I
request u once again plzz send me the config I will be very
greatful. 







Best Regards

Mohammad Zeeshan Latif

Sr. WAN Engineer

NETWORK DIRECTORATE



0092-51-90391020,0092-321-5181157


















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[Asterisk-Users] help about modem

[Muhammad atif amin]

Hi,
 Sorry if question is stupid..As i m just new to asterisk..
I need help in the following schenerio..
Actually i want to transfer incoming call from PSTN to any PC in the LAN. 
Can i use modem for this purpose and also need help in configuration
for this schenerio.woul any one plz give configuration sample
reagarding my problem..

Thanks in advance

Best Regards
___
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[Asterisk-Users] Asterisk Redundancy

[Malik Muhammad]

Hi,
i have two [EMAIL PROTECTED] 2.2 server. i want if one of my asterisk server down. other is taken control of my first server and call goes through.
Is it possible in asterisk.

Usman
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[Asterisk-Users] connection between asterisk and cisco

[muhammad usman]

HI!

how are you people. i am a newbie in asterisk and
voip.
i need your help.

the scenerio is like this.

1.all local SIP users will be connected to asterisk
via IP.

2.PSTN will be connected to AS5300.pstn will give us a
local prefix like 333. so any one calling at
333 will go to my as5300.

3.now i want if someone calls via PSTN to a number
333 this should go to my some sip user e.g john
(connect to asterisk via ip). but only to john.

4.now when john dials to any number outside 333 range
, it should be dialed to the destination via
AS5300(which is connected to PSTN). and destination
should see that it is called by a number 333.

5.now if all this scenerio is possible, how the
asterisk server and As5300 will talk to each other.
what protocol can be used between them.
and what physical connection i.e like ethernet or E-1
connection between as5300 and asterisk server.


6.which billing radius server you recommend, and what
kind of cards will be required in a5300.


thanks a lot for reading this.
and thanks for reply in advance.


any other suggestions are also welcome.

regards







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Re: [Asterisk-Users] [Asterisk-Dev] Asterisk 1.2 Release Plans

[Muhammad Nuzaihan Kamalluddin]

Hi,

He mentions Asterisk 1.2 not [EMAIL PROTECTED] 1.2. You got it confused.


On 7/28/05 7:25 AM, Scott Stone [EMAIL PROTECTED] wrote:

 
 The current version up for download is 1.3  how does that mesh with
 a potential release of 1.2, in the future, when 1.3 is out on
 http://asteriskathome.sourceforge.net already?  Or is the
 asteriskathome project something different?
 
 On Wed, 2005-07-27 at 23:18 +0200, TWV wrote:
 What are all these astonishing new features and improvements?
 Can you please give us an overview?
 
 Thanks!
 
 
 -Oorspronkelijk bericht-
 Van: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Namens Kevin P. Fleming
 Verzonden: dinsdag 26 juli 2005 18:15
 Aan: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
 Developers Mailing List
 Onderwerp: [Asterisk-Users] [Asterisk-Dev] Asterisk 1.2 Release Plans
 
 As previously mentioned on the lists by Olle Johannson, we are actively
 trying to get Asterisk in shape for a 1.2 release within the next 60
 days. To accomplish this, we need a few things to happen:
 
 1) A feature freeze - This will occur at the end of this month, with no
 new feature submissions accepted after July 31st. Any _pending_ feature
 patches in Mantis that have passed architecture review and functionality
 testing before August 1st can be accepted into 1.2, if they make it
 through the remainder of the review processes and are able to be merged
 before August 15th.
 
 2) Progress on open bugs - There are a number of bugs open in Mantis
 that are waiting for the poster to provide additional information, test
 results, call traces, etc. We would much prefer to not release 1.2 with
 suspected problems already identified, but we cannot solve them without
 adequate input from you. If you have an open bug and are not in a
 position to continue providing assistance in solving it, please post a
 message to the mailing lists asking for volunteers to help replicate the
 problem so it can get resolved.
 
 3) Testing - We need a _lot_ of help testing. If you have not previously
 tested CVS HEAD, please download it, read the UPGRADE.txt file and
 install it on one or more systems to play around with. Please do _not_
 put it into a production environment unless you are willing to accept
 the consequences of that action. If you do find a bug or other issue,
 when you open a bug in Mantis, please try to provide _all_ the
 configuration information, call traces, etc. that the bug guidelines
 request, so that we don't waste 3-4 days just going back and forth
 requesting more information from you. If possible, join the #asterisk or
 #asterisk-dev IRC channel to find out exactly what debugging information
 will be required and how to produce it, if you don't already have that
 knowledge.
 
 4) Release Candidates - I will produce the first release candidate on
 August 20th, with followup versions produced every week until we deem
 the release ready for public consumption. I expect it will require at
 least three -RC releases for us to get things in shape, so that means
 that 1.2 itself may be ready by September 15th.
 
 We are very thankful for the community's help and support, and we want
 Asterisk 1.2 to be as important a release as 1.0 itself was. The number
 of new features, performance improvements, bug fixes and
 interoperability enhancements in CVS HEAD is astonishing, and a very
 large percentage of them came directly from community contributions. We
 hope that all of the 'non-developers' in the community will be able to
 help us 'shake out' the bugs and problems remaining in the code, so we
 can be assured of the most stable 1.2 release possible :-)
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[Asterisk-Users] Problem - jittery.

[Muhammad Nuzaihan Kamalluddin]

Hi all,

I am fairly new to asterisk, i have been having this problem for  
the past few weeks. When i press *62 (wake up call), it was working  
fine last time, but now, it goes like, Please enter the time for  
your wake up.. and stops there.


How do i debug this problem?

Thanks,
Zaihan
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Re: [Asterisk-Users] Problems installing asterisk-addons

[Muhammad Nuzaihan Kamalluddin]

try #include asterisk.h in the source. asterisk.h is means that  
it will look up for the file in the same directory.


On 22 Jul 2005, at 4:45 AM, Angus Comber wrote:


I am now getting this make error:

cc -fPIC -I../asterisk -D_GNU_SOURCE  -I/usr/include/mysql -c - 
o cdr_addon_mysql.o cdr_addon_mysql.c

cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory
make: *** [cdr_addon_mysql.o] Error 1
linux:/usr/src/asterisk-addons #

But I have the asterisk sources in /usr/include/asterisk but I am  
installing asterisk-addons from /usr/src/asterisk-addons/ Is that a  
problem?


I think the problem is in line 29 - #include asterisk.h of  
cdr_addon_mysql.c . I assume that I should not really have to edit  
any of the source or make files.  I bet something fairly basic is  
wrong.  any ideas?


Angus




- Original Message - From: Tzafrir Cohen  
[EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Thursday, July 21, 2005 2:55 PM
Subject: Re: [Asterisk-Users] Problems installing asterisk-addons




On Thu, Jul 21, 2005 at 03:44:03PM +0200, Christoph Eicke wrote:


On Thursday 21 July 2005 15:28, Angus Comber wrote:
 My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j

 It is a version put together by Junghanns.net - for working  
with their  ISDN
 cards.  Mmm I wonder if that is the problem?  If so then what  
version  of
 asterisk-addons do I install.  I didn't see anything about   
asterisk-addons

 on the junghanns.net site.

You are right, that is the problem. I wasn't able to compile the  
addons with
the version from junghanns.net. I suspect that it's because those  
addons
compile the MySQL realtime extension and the Asterisk version  
coming with the

bristuff package has no support for the realtime extension yet.



1.0.9 has no support for realtime yet, both in addon in in the main
distribution. You seem to be mixing 1.0 and HEAD.

--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] RE: Asterisk-Users] te405p and dell poweredge

[Muhammad Nasim]

This works fine.


Message: 26
Date: Tue, 7 Jun 2005 17:22:23 +0800
From: Ma Zhiyong [EMAIL PROTECTED]
Subject: [Asterisk-Users] te405p and dell poweredge
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=gb2312

Hi, I want to install * and te405p on Dell Poweredge 1850. Can I do that
successfully? Any one has successful experience on that scenario?
Thanks.
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End of Asterisk-Users Digest, Vol 11, Issue 45
**


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[Asterisk-Users] help regading outbound calls

[Muhammad Haris]

dear fellows,
i  succesfully deployed a voip based call center using Cisco 2600
series routers as gateway and developed my own AGI which is an IVR
application in C language in which i performed database transactions
using PostGreSQL database.
now as per my office requirement i have to setup one local land line
for inbound  outbound calls to PSTN. although inbound calls are made
perfectly but i dont know what is wrong with outbound calls i didnt
able to make any outbound calls through asterisk. i have a TDM400P
card with 1 FXS at signalling port 4 and 1 FXO at signalling port 1.
analog phone is connected to the FXS port while main PSTN line is
connected to FXO port. everything looks perfect instead when i dial a
local number (say 0216998256), asterisk displays a ringing msg then
display that;

 Dial Zap1-1,Zap1 
Everyone is Busy/Congested
hangup.
plz help me and if possible for either of you then plz send me
configurations as well b/c i am in a hurry and i have to setup
everything in justa couple of days...

thanks in advance and thanks for your contemplation.
regards 

haris
:-(
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Re: [Asterisk-Users] setting SIP to dial PSTN with TDM400P

[Muhammad Haris]

to dear martijn,

i made every possible change i can make 
i have a TDM400P Zap card...
i had connected PSTN line to FXO Kewlstart at channel 1.
and analog phone to FXS Kewlstart at Channel 4.
i can hear continous ring tone when i hook up the receiver.
plz have a look at my confs.

my extension.conf is as follows;

[pstn-outbound]
exten = _.,1,Dial(Zap/1/${EXTEN})
exten = _.,2,Congestion

my zaptel.conf is as follows:

 [channels]
;
; Default language
;
language=en
musiconhold=default
usercallerid=yes
hidecallerid=no
callreturn=yes
callprogress=no

rxwink =300
echotraining=800
rxgain=0.0
txgain=0.0

busydetect=1
busycount=7

immediate=no

signalling=fxo_ks

;callerid=asreceived

context=pstn-outbound
channel=1

relaxdtmf=yes
callwaiting=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
;
transfer=yes



when i dial a local number say (6998256) from analog phone set then
asterisk shows following messages.

*CLI -- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/1/6998256) in new stack
-- Called 1/6998256
-- Zap/1/6998256-busy-1013475805 is busy
-- Hungup 'Zap/1/6998256-busy-1013475805'
  == Everyone is busy/congested at this time
-- Timeout on Zap/1-1
  == CDR updated on Zap/1-1

**
please reply with your suggestions i always take care to run ztcfg
command whenever i made any changes to zaptel.conf.plz help me solving
this problem
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[Asterisk-Users] how to call land line number using wireless land line service through asterisk

[Muhammad Haris]

dear fellows,

i want to make a call from an ip phone to a local pstn number (say
021-699-8256) using a Wireless phone service. asterisk comes in b/w
the ip phone and the Wireless phone which is connected to fxs port#1.
i just want to dial a number (above mentioned) through ip-phone and
asterisk connects a call through Wireless service. the wireless
service charge every call i make in this way

please reply me if you made a setup of this kind.. i apologiezes to
all if these kinds of messages already answered before. if it is then
refer me the mail that was answered..
regards.
haris  
Xnet Solutions Inc. Karachi,Paksitan.
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[Asterisk-Users] how to call land line number using wireless land line service through asterisk

[Muhammad Haris]

i dont have anolog phone at my office. anyway i will arrange it for
instance. but can u do me a favour? plz resolve y queries...

i have connected a anolog phone to Fxs port-1 at asterisk machine now
plz send me the configuration of extension.conf to make outbound
calls. i had configured zaptel.conf and zapata.conf.
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[Asterisk-Users] setting SIP to dial PSTN with TDM400P

[Muhammad Haris]

I've setup * with TDM400P w/1 FXS, 1 FXO modules.
I've one analog phone connected to TDM400P FXS module, 1 PSTN line to
one of the FXO module(ZAP) , and IP phone connected to asterisk on
LAN.
 
The calls between SIPs and zap phone (fxs) are OK.  But 2 issues
cannot be solved:
 
1. To dial to PSTN via zap phone, the setup in extensions.conf with
the following
 exten = _Nxx, 1, zap/1
   doesn't work.
   Does anyone can give me suggestion that what did I do wrong to make
the setting:
 
2. When trying using SIP phone to dial PSTN, I got no luck.

   Please advice if any solution.
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[Asterisk-Users] noice sip to sip only???

[Muhammad Muzzamil Luqman]

i have been using the asterisk for some three 
weeks. Previously i was using the softphone iax-phone and now i have to shift to 
the sip phone xlite.

The problem is that there's always unbearable noice 
in sip to sip calls. Is there any way to get rid of this

Kindest
MM Luqman
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[Asterisk-Users] noice sip to sip only???

[Muhammad Muzzamil Luqman]

i have been using the asterisk for some three 
weeks. Previously i was using the softphone iax-phone and now i have to shift to 
the sip phone xlite.

The problem is that there's always unbearable noice 
in sip to sip calls. Is there any way to get rid of this

Kindest
MM Luqman
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[Asterisk-Users] msic while ringing

[Muhammad Muzzamil Luqman]

I want to setup a senario in which the callers 
hears to some music file while the phone is ringing and as soon as the line is 
answered the music is stopped palying. i.e. instead of the rings the caller 
listens to some music.

Is is possible with asterisk?

Kindest
Muhammad Muzzamil Luqman
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[Asterisk-Users] a silly question regarding call monitoring!

[Muhammad Muzzamil Luqman]

i want to use the call monitoring for my litttle 
exchange. As my requirements are to monitor each and every call. Is there any 
way to do this wil a single command or i shall have to write it for each and 
every extension and the incomings and the outgoings.
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[Asterisk-Users] Any $CALLER

[Muhammad Muzzamil Luqman]

I am working on call monitoring.
The filename convention that i am using 
is:

${EXTEN}--- timestamp.wav

i want to add the callers extension to it as well. 
Is there any variable alreadu defined for that?

Kindest
Muhamamd Muzzamil Luqman
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[Asterisk-Users] mixing sound files?

[Muhammad Muzzamil Luqman]

As soon as a call hits the asterisk a menu is 
played "Press 1 for ... and 2 for..."

I have got the speech in different mp3 file and the 
music in different mp3 file. is there anyway to mix these two 
files?

Kindest
Muhammad Muzzamil Luqman
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[Asterisk-Users] asterisk -vvvvvvvgrc?

[Muhammad Muzzamil Luqman]

what does the parameter -vvvgrcmeanand are there any others as well?KindestMuhammad Muzzamil Luqman
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[Asterisk-Users] does asterisk support menus?

[Muhammad Muzzamil Luqman]

Whenever some call comes in i want it to be 
automatically picked up and then it plays some message "Welcome to 
xyz,Press 1 for sales and 2 for support" and then it takes it to the 
particular extension of sales/support.

can i achieve this thing using 
asterisk?

Kindest
Muhammad Muzzamil Luqman 

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[Asterisk-Users] mp3 to gsm?

[Muhammad Muzzamil Luqman]

i have got a music file with extension mp3 and it 
is not workign with background()

is there any way to convert the mp3 to gsm or any 
other codec?

Kindest 
Muhammad Muzzamil Luqman
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[Asterisk-Users] can't enable trunking :(

[Muhammad Muzzamil Luqman]

I have successfully installed and configured the 
asterisk, the incoming and the outgoing calls are working fine, its a tremendous 
solution :)

Now i want to enable trunking between two asterisk 
boxes, in the iax.conf i have put:

[karachi]
...
...
...
trunk=yes
...
...
...

everything seems to work fine but when i load 
asterisk it says:

--
Feb 17 10:59:14 WARNING[18726]: chan_iax2.c:7536 
build_user: Unable to support trunking on user 'karachi' without zaptel 
timingFeb 17 10:59:14 WARNING[18726]: chan_iax2.c:7345 build_peer: Unable to 
support trunking on peer 'karachi' without zaptel timing
--

I tried to install the ztdummy and i succeeded on 
one of the box but for the other i am having problems :(

It was missing the kernel-source rpm. I installed 
the version that i found but now the first error is still there and when i 
modprobe ztdummy it gives the following response.
---
[EMAIL PROTECTED] asterisk]# modprobe 
ztdummy/lib/modules/2.4.25-040218/misc/zaptel.o: kernel-module version 
mismatch 
/lib/modules/2.4.25-040218/misc/zaptel.o was compiled for kernel version 
2.4.20-24.9 while this kernel is 
version 2.4.25-040218./lib/modules/2.4.25-040218/misc/zaptel.o: insmod 
/lib/modules/2.4.25-040218/misc/zaptel.o 
failed/lib/modules/2.4.25-040218/misc/zaptel.o: insmod ztdummy 
failed[EMAIL PROTECTED] asterisk]# 
--


Any Kind peace of information will be highly 
appriciated :)

Best Regards
Muhammad Muzzamil 
Luqman
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