from:"Asghar Mohammad"

Re: [asterisk-users] Dialing a call back out on same SIP trunk as it came in

2015-11-25 Thread Asghar Mohammad

I had in a same situation and solved by Background 1 sec. silence.

On Wed, Nov 25, 2015 at 5:45 PM, Brian ::  wrote:

> add a pause in the dialplan for a second then proceed..
>
>
>
> On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield 
> wrote:
>
>> In article <20151125133008.6369360.14455.17...@gmail.com>,
>> Israel Gottlieb  wrote:
>> > Try putting progress instead of answer
>>
>> Yes, I tried Progress already, and it didn't help. But thanks for
>> the suggestion!
>>
>> Tony
>>
>> > I have a puzzling situation, and would be grateful for any insight.
>> >
>> > I have a dialplan that forwards an incoming call out to another
>> > number via the same SIP trunk as it came in on. e.g.
>> >
>> > [from-siptrunk]
>> > exten => 0123456789,1,NoOp
>> > exten => 0123456789,n,Dial(SIP/siptrunk/0987654321)
>> >
>> > Now, if I use a different SIP trunk for the outbound call, than the
>> > inbound call came on, the call is set up fine - the Answer signal from
>> the
>> > called party gets propagated back to the caller, and they can hear each
>> > other.
>> >
>> > But if the outbound SIP trunk is the same as the one the call came in
>> on,
>> > the caller doesn't hear any progress, and has no notification of when
>> the
>> > call was answered. Neither can the parties hear each other.
>> >
>> > I have tried this on two different machines using two different SIP
>> > providers.
>> >
>> > However, if I change the above NoOp to be Answer(100), i.e. answer the
>> > inbound call before placing the outbound Dial, the caller hears progress
>> > and when the called party answers, they hear each other fine.
>> >
>> > Of course, if the called party is busy, the caller just hears in-band
>> > busy tone, as the caller's inbound call was already answered.
>> >
>> > Can anyone explain why I need the Answer? It feels wrong that I should.
>> >
>> > The siptrunk entry contains canreinvite=no and directmedia=no.
>> >
>> > The version of Asterisk on these boxes is 10.5.1, if that's relevant.
>> >
>> > Thanks for any insight!
>> >
>> > Cheers
>> > Tony
>> >
>> > --
>> > Tony Mountifield
>> > Work: t...@softins.co.uk - http://www.softins.co.uk
>> > Play: t...@mountifield.org - http://tony.mountifield.org
>> >
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>>
>>
>> --
>> Tony Mountifield
>> Work: t...@softins.co.uk - http://www.softins.co.uk
>> Play: t...@mountifield.org - http://tony.mountifield.org
>>
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>
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Re: [asterisk-users] Calls not hanging up

2014-08-07 Thread Asghar Mohammad

Your call is up on VoiceMail you should check dialstatus before sending
user to VoiceMail.


On Thu, Aug 7, 2014 at 4:12 PM, D'Arcy J.M. Cain da...@vex.net wrote:

 This just started after upgrading to 11.11.0.  After a call is
 completed (both ends hang up) the call still shows as active.

 # asterisk -x core show channels
 Channel  Location State   Application(Data)
 SIP/thinktel-000 (None)   Up  AppDial((Outgoing
 Line)) SIP/4164251212-0 416555@LocalSets Up
 Dial(SIP/thinktel/416555) 2 active channels
 1 active call
 1 call processed

 The 1212 number is mine and is hung up.  I even rebooted my ATA to make
 sure that it wasn't holding the line.  My dialplan is extremely
 simple.  In fact, I even simplified it from what it was for this
 testing.  Here it is.

 exten = 4164251212,1,Verbose(0, ${CALLERID(all)} Calling ${EXTEN})
 same = n,Dial(SIP/4164251212,30)
 same = n,VoiceMail(4164251212@LocalSets,u)
 same = n,Hangup()

 I can post any other log or config excerpts if someone thinks that they
 are relevant but all of this was working under 11.10.2.

 Thanks.


 --
 D'Arcy J.M. Cain
 System Administrator, Vex.Net
 http://www.Vex.Net/ IM:da...@vex.net
 VoIP: sip:da...@vex.net

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Re: [asterisk-users] dialplan =how many concurrent calls

2014-07-10 Thread Asghar Mohammad

you can use GROUP and GROUP_COUNT

n,Set(GROUP()=aname)
n,GotoIf($[${GROUP_COUNT(aname)}  8]?${EXTEN},200)
200,Hangup


On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser visser.raf...@gmail.com
wrote:

 Hi guys.

 Does somebody knows how to get the concurrent calls from the dial plan?

 Or.

 How can i control not to run more than n simultaneus agi applications?

 Thanks in advance.
 rv

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Re: [asterisk-users] Trunk issue

2014-04-28 Thread Asghar Mohammad

file is executable?
can you show ls -l /var/lib/asterisk/agi-bin


On Mon, Apr 28, 2014 at 7:12 PM, Haley,Scott A
scott.ha...@edwardjones.comwrote:

 It runs but hangs with the output of:
 perl tbsdial.agi 81101
 GET VARIABLE astexten


 Right now, it is a simple perl script. Here is the entire script.

 #!/usr/bin/perl


 use Asterisk::AGI;

 my $agi = new Asterisk::AGI;

 my $dialgroup1 = DIALGROUP1;
 my $dialgroup2 = DIALGROUP2;
 my $vmvariable = VM;
 my $timer = TIMER;
 my $branch = BRANCH;
 my $input;
 my $dg1value;
 my $dg2value;
 my $vmvalue;
 my $branchvalue;



 $input = $agi-get_variable(astexten);

 #$agi-answer();
 #$agi-stream_file(welcome);






 $agi-set_variable($dialgroup1, $dg1value);
 $agi-set_variable($dialgroup2, $dg2value);
 $agi-set_variable($vmvariable, $vmvalue);
 $agi-set_variable($timer, $timervalue);
 $agi-set_variable($branch, $branchvalue);

 Thanks,
 Scott Haley
 5-2244





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 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
 Sent: Monday, April 28, 2014 12:07 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue

 Does the script generate an error when run outside of Asterisk?   An AGI
 should simply wait for input when run outside of Asterisk.


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
 Sent: Monday, April 28, 2014 1:04 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue

 I am trying to run an agi script and asterisk is not finding it. The
 output of the cli is as follows:

 -- Executing [s@tbs-utils:7] AGI(SIP/7002-001a, tbsdial.agi) in
 new stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681
 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi':
 File does not exist.

 The file is in that directory and is owned by the user asterisk. Why
 does it say the file does not exist?
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Re: [asterisk-users] Trunk issue

2014-04-28 Thread Asghar Mohammad

if that is the case then check again Perl Asterisk AGI.


On Mon, Apr 28, 2014 at 7:33 PM, Haley,Scott A
scott.ha...@edwardjones.comwrote:

 One more thing. I have this exact same script working on an Asterisk 1.8
 box. This is a new Asterisk 11.7 box.

 Thanks,
 Scott Haley
 5-2244


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
 Sent: Monday, April 28, 2014 12:32 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue

 Here is the directory listing:

 [root@nxdasterisk-3 agi-bin]# ls -al
 total 12
 drwxr-xr-x.  2 asterisk asterisk 4096 Apr 28 12:11 .
 drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
 -rwxrwxr-x.  1 asterisk asterisk  590 Apr 28 11:55 tbsdial.agi

 Thanks,
 Scott Haley
 5-2244


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
 Sent: Monday, April 28, 2014 12:25 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue


 Odd.  AGI scripts should hang waiting for input when run from the CLI.
  They should not output anything.  If the script is not set as executable
 you'd get an error.

 If you were not running it as the same user as asterisk runs as you should
 still get a different error.


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
 Sent: Monday, April 28, 2014 1:13 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue

 It runs but hangs with the output of:
 perl tbsdial.agi 81101
 GET VARIABLE astexten


 Right now, it is a simple perl script. Here is the entire script.

 #!/usr/bin/perl


 use Asterisk::AGI;

 my $agi = new Asterisk::AGI;

 my $dialgroup1 = DIALGROUP1;
 my $dialgroup2 = DIALGROUP2;
 my $vmvariable = VM;
 my $timer = TIMER;
 my $branch = BRANCH;
 my $input;
 my $dg1value;
 my $dg2value;
 my $vmvalue;
 my $branchvalue;



 $input = $agi-get_variable(astexten);

 #$agi-answer();
 #$agi-stream_file(welcome);






 $agi-set_variable($dialgroup1, $dg1value);
 $agi-set_variable($dialgroup2, $dg2value);
 $agi-set_variable($vmvariable, $vmvalue); $agi-set_variable($timer,
 $timervalue); $agi-set_variable($branch, $branchvalue);

 Thanks,
 Scott Haley
 5-2244





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 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
 Sent: Monday, April 28, 2014 12:07 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue

 Does the script generate an error when run outside of Asterisk?   An AGI
 should simply wait for input when run outside of Asterisk.


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
 Sent: Monday, April 28, 2014 1:04 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue

 I am trying to run an agi script and asterisk is not finding it. The
 output of the cli is as follows:

 -- Executing [s@tbs-utils:7] AGI(SIP/7002-001a, tbsdial.agi) in
 new stack [Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681
 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi':
 File does not exist.

 The file is in that directory and is owned by the user asterisk. Why
 does it say the file does not exist?
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Re: [asterisk-users] FW: clients unable to auth

2014-04-16 Thread Asghar Mohammad

Hello,
Try this

[6004]

type=friend

host=dynamic

disallow=all

allow=ulaw

allow=alaw

callerid=6004 Peter

secret=XXX

context=default

port=9060

nat=force_rport,comedia

deny=0.0.0.0

permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0





On Wed, Apr 16, 2014 at 12:56 PM, Peter Reid peter.r...@morodo.co.ukwrote:

 Hi Guys,



 Just new to Asterisk and am completely stumped.  I have created two
 accounts as instructed.  Please see below for the config of the user
 accounts.



 [Peter]

 type=friend

 host=IP address

 disallow=all

 allow=ulaw

 allow=alaw

 callerid=Peter 6004

 secret=XXX

 context=default

 port=9060

 nat=force_rport,comedia

 deny=0.0.0.0

 permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0



 When attempting to register there appears to be something not allowing the
 authentication of the client against Asterisk.  I am getting a 401
 Unauthorized on first attempt and then 403 (Bad auth) on second.   Is there
 any ACL config that I have missed which is not allow a good
 authentication.  I have enabled nat and allowed all public and private IP’s
 of the two clients with masks.



 The CLI console returns the following:



 ignore - Apr 16 11:09:57] NOTICE[24359]: chan_sip.c:27724
 handle_request_register: Registration from '6004
 sip:6...@xx.xx.xx.xx:9060' failed for 'IP:57836' - No matching peer found

 [Apr 16 11:09:57] NOTICE[24359]: chan_sip.c:27724 handle_request_register:
 Registration from '6004sip:6004@IP:9060' failed for 'IP:57836' - No
 matching peer found

 serverIP*CLI



 Have searched under No matching peer found and although there is some info
 on this it does not appear to satisfy my situation.



 Please help.



 *Best Regards, *



 *Peter *








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Re: [asterisk-users] Asterisk 1.8.24 : illegal instruction

2013-11-20 Thread Asghar Mohammad

Hello,
you can check the asterisk binary with.
file /usr/sbin/asterisk
and linked library
ldd /usr/sbin/asterisk




On Wed, Nov 20, 2013 at 2:51 PM, Jonas Kellens jonas.kell...@telenet.bewrote:

  On 20-11-13 14:43, A J Stiles wrote:

 On Wednesday 20 November 2013, Jonas Kellens wrote:

  Hello,

 I have installed asterisk 1.8.24 (from source) but I can not start up
 Asterisk :


 [root@sip32 admin]# /usr/sbin/asterisk -r
 Illegal instruction

  Are you using a VIA C6/C7 processor  (often found soldered to tiny
 motherboards),  by any chance?  This family of processors falsely report as
 i686 when they lack some of the instructions for this family.

 The fix is to build for a target architecture of i586.



 No, this is a Xen VPS.



 Jonas.


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Re: [asterisk-users] outbound call issue

2013-10-18 Thread Asghar Mohammad

some more information's will help sort out the issue.


On Fri, Oct 18, 2013 at 2:30 PM, shiva kumar sivakumar.kara...@gmail.comwrote:

 Dear All,


 i had an issue when we are going to call back the number from asterisk
 its ringing as the customer mobile is switched off.
 And also it also not saying busy when the customer is on another
 call.

 so please help me in this issue

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Re: [asterisk-users] MusicOnHold starts magically for no reason

2013-10-18 Thread Asghar Mohammad

if you don't use MOH just don't load module res_musiconhold.so


On Fri, Oct 18, 2013 at 6:24 PM, Alban Elziere alban.elzi...@nevox.frwrote:

 Thank you for pointing this thread.
 So, looks like no solution exists to correct this (as I understand)... as
 it is part of the standard. Have you found a trick to avoid that (break it)?

 Alban

 -Message d'origine-
 De : asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] De la part de Doug Lytle
 Envoyé : vendredi 18 octobre 2013 13:55
 À : Asterisk Users Mailing List - Non-Commercial Discussion
 Objet : Re: [asterisk-users] MusicOnHold starts magically for no reason

  I also see that on our servers. By the way, is It possible to avoid
 this behavior?  It's quite disappointing for our customers to hear their
 music on hold  when the remote party put them on hold...

 You'll want to review  this thread:

 http://www.asteriskguru.com/archives/image-vp345921.html

 Doug

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Re: [asterisk-users] What linux distro most popular for Asterisk

2013-10-17 Thread Asghar Mohammad

We are using Debian 32bit and 64bit on standalone and on VMs without any
issue.



On Thu, Oct 17, 2013 at 10:15 AM, Frederic Van Espen
frederic...@gmail.comwrote:

 On 10/17/2013 09:47 AM, Alban Elziere wrote:

 I'm using Ubuntu server (32bit mainly), standalone or VM (esxi) with good
 stability.


 Same here. We've been using ubuntu lucid 32bit for years. We have about
 1000 implementations of this.

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Re: [asterisk-users] is g729 codec free? or under license???

2013-10-14 Thread Asghar Mohammad

Hi,
no you don't need just allow g729 on both peers or allow all on both peers
and enable only g729 on softphones.
my asterisk in middle don't have g729 and g723.1 on my asterisk but my
both end point have these codecs i just allowed the codes and it works
perfectly.

On Mon, Oct 14, 2013 at 9:21 AM, s m sam.gh1...@gmail.com wrote:

thanks Asghar, but are you sure? my two endpoints -which are soft-phones-
have g729 codec but my asterisk on middle system has not any module for
g729 codec. i think i should get module g729 for my middle system in order
to pass calls with g729 codec. isn't it true?

On Sat, Oct 12, 2013 at 1:08 PM, Asghar Mohammad asghar...@gmail.comwrote:

HI,
You don't need a g729 installed in pass throw mode. if both ends have
codec g729 you can just enable on both peers.
and asterisk should pass the codec from 1 end to other.
but make sure you are not doing transcoding of any type answering the
call playing voice prompts etc.

On Sat, Oct 12, 2013 at 9:52 AM, s m sam.gh1...@gmail.com wrote:

thank you everybody for your useful replies and so sorry to answer late.

i understand what i need. first of all, i wanna to use pass through g729
codec (which is free). so i go to http://asterisk.hosting.lv/ to get
g729 codec. i have freebsd 8.2 and asterisk 1.8.22 but there is no
compatible codec for Xeon Intel in the list. it means that i can't use
codec g729 on my system??? or can i use codec for another type of hardware
for my system? anyone has any experience?

thanks in advance
SAM

On Mon, Oct 7, 2013 at 5:04 PM, John Novack
jnov...@stromberg-carlson.org wrote:

Darryl Moore wrote:

Thank you Steve, and I read a bit more on the web on this subject
including your own well reasoned page at
http://www.soft-switch.org/**patents/index.htmlhttp://www.soft-switch.org/patents/index.html

However, despite wide acceptance of the patentability of such codecs
(unfortunately), whether they are in fact software patents or not
appears to be a matter of opinion. The FSF and Fedora both refer to
codec patents as being software patents.

http://endsoftpatents.org/**2011/02/usa-patent-reform-not-**enough/http://endsoftpatents.org/2011/02/usa-patent-reform-not-enough/
http://fedoraproject.org/wiki/**Software_Patentshttp://fedoraproject.org/wiki/Software_Patents

A quick google search of both terms will show that there are a great
many people who see codec patents as software patents, so I don't think
I am alone there.

snip

Law is ALWAYS open to interpretation, so that is not surprising.
See if you can get any lawyer, and especially a patent attorney, to
give you a definitive answer! You will not get one.
Seldom will you ever get an eggspurt legal opinion Any good lawyer
will tell you maybe, or if there is any doubt don't do it!
Law is not precisely measurable. No meter or O'scope to assist here.
Any A**hole can sue anyone for the filing fee, and the results are up
to the opinion of a judge or jury.
The lawyers want it that way, so it isn't ever going to be any
different.

John Novack

Dog is my Co-pilot

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Re: [asterisk-users] is g729 codec free? or under license???

2013-10-12 Thread Asghar Mohammad

HI,
You don't need a g729 installed in pass throw mode. if both ends have codec
g729 you can just enable on both peers.
and asterisk should pass the codec from 1 end to other.
but make sure you are not doing transcoding of any type answering the call
playing voice prompts etc.

On Sat, Oct 12, 2013 at 9:52 AM, s m sam.gh1...@gmail.com wrote:

thank you everybody for your useful replies and so sorry to answer late.

thanks in advance
SAM

On Mon, Oct 7, 2013 at 5:04 PM, John Novack jnov...@stromberg-carlson.org
wrote:

Darryl Moore wrote:

Thank you Steve, and I read a bit more on the web on this subject
including your own well reasoned page at
http://www.soft-switch.org/**patents/index.htmlhttp://www.soft-switch.org/patents/index.html

A quick google search of both terms will show that there are a great
many people who see codec patents as software patents, so I don't think
I am alone there.

snip

Law is ALWAYS open to interpretation, so that is not surprising.
See if you can get any lawyer, and especially a patent attorney, to give
you a definitive answer! You will not get one.
Seldom will you ever get an eggspurt legal opinion Any good lawyer will
tell you maybe, or if there is any doubt don't do it!
Law is not precisely measurable. No meter or O'scope to assist here.
Any A**hole can sue anyone for the filing fee, and the results are up to
the opinion of a judge or jury.
The lawyers want it that way, so it isn't ever going to be any different.

John Novack

Dog is my Co-pilot

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Re: [asterisk-users] Capture Media IP in CDR

2013-10-12 Thread Asghar Mohammad

hi,
you have not mentioned which cdr backend you are using.
peer ip is saved in variable CHANNEL(peerip).
if you are using mysql for cdr backend you can create a field in cdr table
(field name can b any of your choice)
in dialplan assign the value of CHANNEL(peerip) to you ip field and
asterisk will fill ip field.
if you name ip feild Peer_ip you can use this example.

same,n,Set(CDR(Peer_ip)=${CHANNEL(peerip)})




On Sat, Oct 12, 2013 at 4:05 AM, CDR vene...@gmail.com wrote:

 I am not proxying the media, but never the less I am forced to store
 the source media IP in my CDR, for regulatory reasons. Asterisk gets
 that information when the reinvite comes, but how do I store it?
 If I don't figure this out my next email will be from Federal Prison.
 Kindly help me stay away from those guys. Eventually we all need to
 save that information or we shall not be able to stay in business.

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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-01 Thread Asghar Mohammad

Hi,
Bad boys trying to guess a valid username.
in sip.conf uncomment  alwaysauthreject=yes and Asterisk always reject 1st
invite.


On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades 
mailinglist+aster...@dns99.co.uk wrote:

  On 01/10/13 15:44, gincantalupo wrote:

 On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo gincantal...@fgasoftware.com
  wrote:

 Hi,

 I get a lot of these messages on my Asterisk CLI:

 Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS
 ;tag=03f82bb9

 as if my PBX machine is trying to authenticate to itself. It seems
 someone is attacking my asterisk PBX.

 Is there a way to fix this problem?


 in sip.conf I have guest connections permitted and have them going to the
 default context which contains :-

 [default]
 ; all unauthenticated connection attempts from the internet come in here.
 exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
 ${SIP_HEADER(Contact)})
 exten = _[+*#0-9].,n,Congestion

 Then in fail2ban I have it match the following :-

 failregex = Registration from .* failed for \'HOST\' - Wrong password
 Unauthenticated call attempt .*\@HOST\:


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Re: [asterisk-users] problem to get MWI working

2013-09-29 Thread Asghar Mohammad

HI Asmaa,
I don't know how MWI works in Voicemail but as i understand it just create
a .call file and put in /var/spool/asterisk/outgoing and asterisk execute
that file.
i am using similar method for sending fax to from email.

i show you some examples from my php scripts.

1. in voicemail   context
exten =_X.1,VoicemailMain()
exten =h,1,exten = h,n,System(/usr/bin/php path to php script
 ${CDR(accountcode)} ${FAXEDNUM} ${CALLERID(num)} ${FAXSTATUS}
${CDR(duration)} ${UNIQUEID} ${CALLCOUNT} ${TIFF})

pass variables to script as you need.
**Part of php
2. script do some checks on saved VM of Fax if it is too short or
incomplete just delete it or do want you want.
in script collect variables pass to it.
 $argv;
$accountcode = $argv[1];
$callednum = $argv[2];
$callerid = $argv[3];
$faxstatus = $argv[4];
$billtime = $argv[5];
$unid = $argv[6];
$callcount = $argv[7];
$faxtiff = $argv[8];

Create .call file somewhere but not in /var/spool/asterisk/outgoing

$filename = path to call file/$accountcode-$unid.call;

Remove old call file with same if any.

system (rm -f $filename);

Create Contents of file as you need

$Content = Channel: $providertech/$callednum@$providerip\nCallerID:
$callerid\nWaitTime: 180\nMaxRetries: 0\nRetryTime: 300\nContext:
fax-out\nExtension: $callednum \nArchive: false\nPriority: 1\nSetVar:
SENDER=$callerid \nSetVar: TIFF=$faxtiff \nAccount: $accountcode \nSetVar:
CALLCOUNT=2;

Open file and fill it.

$handle = fopen($filename, 'x+');
fwrite($handle, $Content);
fclose($handle);

if you want execute call file after some delay change timestamp.

system (touch -d '3 minutes 11 seconds' $filename);

Move file to /var/spool/asterisk/outgoing

Note: don't copy the file but MOVE the file. if you copy the file asterisk
may execute partial file.

system (/bin/mv $filename /var/spool/asterisk/outgoing);

you can use any scripting language.

Hope this will help you





On Sun, Sep 29, 2013 at 2:01 AM, Asmaa Ahmed asabatg...@hotmail.com wrote:

 Hello,

 It looks that I got this in the logs while running the scripts manually by
 mistake, so back to the starting point I can't figure why externnotify
 doesn't run? My target is  to have MWI (Message waiting indicator)
 running.
 Also still can see the debug logs in CLI/asterisk logs even with
 increasing the verbosity and debug level!

 Thanks.

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Re: [asterisk-users] problem to get MWI working

2013-09-29 Thread Asghar Mohammad

Hi Asmaa,
Have you enabled debug to console in logger.conf?
enable debug in logger.conf console = notice,warning,error,debug and
reload Asterisk.


On Sun, Sep 29, 2013 at 4:48 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:

 Hi Asghar,

 Thanks a lot for your proposed solution!
 MWI is turned on or off by the presence of a msgxxx.txt file in the INBOX
 directory for a given voicemail box. The externnotify= option in
 voicemail.conf allows to run a program or script whenever a voicemail is
 received and also when someone exits the VoiceMailMain() application. When
 externnotify is processed it passes the context, extension and number of
 messages to the program or script you specify.

 My problem is that I don't understand why I can't get it activated
 (externnotify). I don't see it being called at all!
 My second issue is that I don't see debug logs even with increasing the
 verbosity and debugging to 10, Is there something else needed to be done. I
 can only see in Asterisk logs warning and notice levels!

 Thanks.

 --
 Date: Sun, 29 Sep 2013 12:41:55 +0200
 From: asghar...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] problem to get MWI working


 HI Asmaa,
 I don't know how MWI works in Voicemail but as i understand it just create
 a .call file and put in /var/spool/asterisk/outgoing and asterisk execute
 that file.
 i am using similar method for sending fax to from email.

 i show you some examples from my php scripts.

 1. in voicemail   context
 exten =_X.1,VoicemailMain()
 exten =h,1,exten = h,n,System(/usr/bin/php path to php script
  ${CDR(accountcode)} ${FAXEDNUM} ${CALLERID(num)} ${FAXSTATUS}
 ${CDR(duration)} ${UNIQUEID} ${CALLCOUNT} ${TIFF})

 pass variables to script as you need.
 **Part of php
 2. script do some checks on saved VM of Fax if it is too short or
 incomplete just delete it or do want you want.
 in script collect variables pass to it.
  $argv;
 $accountcode = $argv[1];
 $callednum = $argv[2];
 $callerid = $argv[3];
 $faxstatus = $argv[4];
 $billtime = $argv[5];
 $unid = $argv[6];
 $callcount = $argv[7];
 $faxtiff = $argv[8];

 Create .call file somewhere but not in /var/spool/asterisk/outgoing

 $filename = path to call file/$accountcode-$unid.call;

 Remove old call file with same if any.

 system (rm -f $filename);

 Create Contents of file as you need

 $Content = Channel: $providertech/$callednum@$providerip\nCallerID:
 $callerid\nWaitTime: 180\nMaxRetries: 0\nRetryTime: 300\nContext:
 fax-out\nExtension: $callednum \nArchive: false\nPriority: 1\nSetVar:
 SENDER=$callerid \nSetVar: TIFF=$faxtiff \nAccount: $accountcode \nSetVar:
 CALLCOUNT=2;

 Open file and fill it.

 $handle = fopen($filename, 'x+');
 fwrite($handle, $Content);
 fclose($handle);

 if you want execute call file after some delay change timestamp.

 system (touch -d '3 minutes 11 seconds' $filename);

 Move file to /var/spool/asterisk/outgoing

 Note: don't copy the file but MOVE the file. if you copy the file asterisk
 may execute partial file.

 system (/bin/mv $filename /var/spool/asterisk/outgoing);

 you can use any scripting language.

 Hope this will help you





 On Sun, Sep 29, 2013 at 2:01 AM, Asmaa Ahmed asabatg...@hotmail.comwrote:

 Hello,

 It looks that I got this in the logs while running the scripts manually by
 mistake, so back to the starting point I can't figure why externnotify
 doesn't run? My target is  to have MWI (Message waiting indicator)
 running.
 Also still can see the debug logs in CLI/asterisk logs even with
 increasing the verbosity and debug level!

 Thanks.

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Re: [asterisk-users] iax: unable to transfer - one way audio

2013-09-28 Thread Asghar Mohammad

Hi,
If you post your configuration someone may help you.


On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy seandar...@gmail.com wrote:

 On 09/27/2013 09:08 PM, Sean Darcy wrote:

 We have zoiper connected over iax to asterisk in Sydney. The call is to
 asterisk in New York. The caller in NZ can hear clearly. Nothing in NY.

 Here's the sydney server:

 -- Accepting AUTHENTICATED call from zoiperipaddr:
  requested format = speex,
  requested prefs = (),
  actual format = ulaw,
  host prefs = (silk16|ulaw|gsm|g722),
  priority = mine
  -- Executing [8447@nz-in:1] Dial(IAX2/n4-270, IAX2/sydney) in
 new stack
  -- Called IAX2/sydney
  -- Call accepted by nyipaddr (format ulaw)
  -- Format for call is (ulaw)
  -- IAX2/sydney-8819 is ringing
  -- IAX2/sydney-8819 answered IAX2/n4-270
  -- Channel 'IAX2/n4-270' unable to transfer
  -- Channel 'IAX2/sydney-8819' unable to transfer
  -- Channel 'IAX2/sydney-8819' unable to transfer
  -- Channel 'IAX2/sydney-8819' unable to transfer

 The NY server:

 -- Accepting AUTHENTICATED call from sydneyipaddr:
  -- requested format = ulaw,
  -- requested prefs = (ulaw|silk16|gsm|g722),
  -- actual format = ulaw,
  -- host prefs = (ulaw|gsm|g722),
  -- priority = mine
  -- Executing [s@incoming-nz:1] Goto(IAX2/home-2152,
 incoming,s,nz-in) in new stack
  -- Goto (incoming,s,5)
  -- Executing [s@incoming:5] Dial(IAX2/home-2152,
 DAHDI/g0SIP/250SIP/251,60,**tT) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
  -- Called DAHDI/g0
  -- Called SIP/250
  -- Called SIP/251
  -- DAHDI/1-1 is ringing
  -- SIP/251-001d is ringing
  -- SIP/250-001c is ringing
  -- DAHDI/1-1 is ringing
  -- DAHDI/1-1 answered IAX2/home-2152
  -- Channel 'IAX2/home-2152' unable to transfer
  -- Hanging up on 'DAHDI/1-1'

 Any help appreciated.

 sean



 FWIW, sydney server is 11.5.1, ny server 11.6.0-rc1.

 sean


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Re: [asterisk-users] How to bind to ipv4 ipv6

2013-09-27 Thread Asghar Mohammad

Hi,
Please Search the List there is already a post and solution.



On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za
 wrote:

 Hi All,

 This is my 1st post so lets go.

 What I need to achieve is the following. I have server with both IPv4
 addresses and IPv6 addresses. The problem that I am encountering is that
 in the sip.conf I am having difficulties to bind to both the IPv4 and
 IPv6 addresses.

 Can someone please assist me in this regard as I need to connect another
 server to this server on IPv6 while the rest of the clients are
 connecting on IPv4.

 I need to know how to get this working?

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Re: [asterisk-users] The call is established but without exchanged voice packets

2013-09-20 Thread Asghar Mohammad

Hello,
If Asterisk version is  1.6 use nat=force_rport,comedia


On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:

 Hello,

 I have set the direct media to be off, but still doesn't work. I am not
 sure about NAT configuration!

 SIP.conf, [general] section
 context=internal
 allowguest=no
 allowoverlap=no
 transport=udp
 bindport=5060
 bindaddr=0.0.0.0
 directmedia=no
 srvlookup=no
 disallow=all
 allow=ulaw
 alwaysauthreject=yes
 canreinvite=no
 nat=yes
 session-timers=refuse
 externip=IP
 localnet=172.16.0.255/255.255.255.0

 The error messages

 [Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728
 handle_request_subscribe: Received SIP subscribe for peer without mailbox:
 7002
 [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt:
 Retransmission timeout reached on transmission
 OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response)
 -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 Packet timed out after 32000ms with no response
 [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up
 call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical
 packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 ).
 [Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The
 canary is no more.  He has ceased to be!  He's expired and gone to meet his
 maker!  He's a stiff!  Bereft of life, he rests in peace.  His metabolic
 processes are now history!  He's off the twig!  He's kicked the bucket.
  He's shuffled off his mortal coil, run down the curtain, and joined the
 bleeding choir invisible!!  THIS is an EX-CANARY.  (Reducing priority)


 Thanks.

 --
 Date: Thu, 19 Sep 2013 13:14:59 +0500
 From: msalman...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] The call is established but without
 exchanged voice packets

 Choose suitable NAT settings from sip.conf

 turn direct media in sip.conf or per peer off


 On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.comwrote:

 Hello,

 I am trying to make my first call on Asterisk to succeed. I have Asterisk
 1.8.10.1 running on Ubuntu machine.
 The configuration is quite simple just for my first test, Trying to have a
 call between two X-lite sipphone. The subscribers succeeded to register and
 the call is established, but still no voice can be heard, and lead the
 call to be disconnected after! By checking the logs, I can see this
 chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on
 transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2
 (Critical Response)

 Here's my  simple sip configuration
 [general]
 context=internal
 allowguest=no
 allowoverlap=no
 bindport=5060
 bindaddr=0.0.0.0
 srvlookup=no
 disallow=all
 allow=ulaw
 alwaysauthreject=yes
 canreinvite=no
 nat=yes
 session-timers=refuse
 externip=IP

 [7001]
 type=friend
 host=dynamic
 secret=123
 context=internal

 [7002]
 type=friend
 host=dynamic
 secret=456
 context=internal

 A snoop capture  for my call is uploaded in the following link. I wonder
 if there is any missing configuration or plugin need to be set here!

 http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992
  
 http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992
 Thanks.


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 _
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 **
 Muhammad Salman
 ***


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Re: [asterisk-users] The call is established but without exchanged voice packets

2013-09-20 Thread Asghar Mohammad

Hello,
i think your logic is wrong please explain me what are you trying to do?
[internal]
exten = 7002,1,Answer()
exten = 7002,n,Playback(vm-nobodyavail)
exten = 7002,n,Hangup()

exten = 7001,1,Dial(SIP/7001,60)
exten = 7001,n,Hangup()

try this dial 7002 and you should listen vm-nobodyavail or 7001 to 7001
extension.


On Fri, Sep 20, 2013 at 4:31 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:

 Hello,

 Here is my  extension context,

 [internal]
 exten = 7001,1,Answer()
 exten = 7001,2,Dial(SIP/7001,60)
 exten = 7001,3,Playback(vm-nobodyavail)
 exten = 7001,4,VoiceMail(7001@main) ;forward to voicemail mailbox
 exten = 7001,5,Hangup()

 exten = 7002,1,Answer()
 exten = 7002,2,Dial(SIP/7002,60)
 exten = 7002,3,Playback(vm-nobodyavail)
 exten = 7002,4,VoiceMail(7002@main)
 exten = 7002,5,Hangup()

 exten = 7003,1,Answer()
 exten = 7003,2,Dial(SIP/7003,60)
 exten = 7003,3,Playback(vm-nobodyavail)
 exten = 7003,4,VoiceMail(7003@main)
 exten = 7003,5,Hangup()

 exten = 8001,1,VoicemailMain(7001@main) ;voicemail retreival
 exten = 8001,2,Hangup()

 exten = 8002,1,VoicemailMain(7002@main)
 exten = 8002,2,Hangup()

 --
 Date: Fri, 20 Sep 2013 16:25:42 +0200
 From: asghar...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] The call is established but without
 exchanged voice packets

 Hello,
 paste you extension context.


 On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed asabatg...@hotmail.comwrote:

 Hello,

 I have Asterisk 1.8.10.1
 Moving to nat=force_rport,comedia hasn't solved the problem. Still having
 the same error!

 I am not sure if this is related to the problem here, but I was trying to
 test my voicemail and got this error No audio available).
 [Sep 20 14:05:41] WARNING[11424]: app_dial.c:2218 dial_exec_full: Unable
 to create channel of type 'SIP' (cause 20 - Unknown)
 [Sep 20 14:05:54] WARNING[11424]: app.c:855 __ast_play_and_record: No
 audio available on SIP/7001-0001??
 [Sep 20 14:06:13] WARNING[11387]: chan_sip.c:3641 retrans_pkt:
 Retransmission timeout reached on transmission
 ZjJkNTY0YzZjMTcwNzcwYTg0NWRiMjlhYzQ4ZjFkOTc for seqno 2 (Critical Response)
 -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions


 Thanks.

 --
 Date: Fri, 20 Sep 2013 16:05:35 +0200
 From: asghar...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] The call is established but without
 exchanged voice packets

 Hello,
 If Asterisk version is  1.6 use nat=force_rport,comedia


 On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.comwrote:

 Hello,

 I have set the direct media to be off, but still doesn't work. I am not
 sure about NAT configuration!

 SIP.conf, [general] section
 context=internal
 allowguest=no
 allowoverlap=no
 transport=udp
 bindport=5060
 bindaddr=0.0.0.0
 directmedia=no
 srvlookup=no
 disallow=all
 allow=ulaw
 alwaysauthreject=yes
 canreinvite=no
 nat=yes
 session-timers=refuse
 externip=IP
 localnet=172.16.0.255/255.255.255.0

 The error messages

 [Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728
 handle_request_subscribe: Received SIP subscribe for peer without mailbox:
 7002
 [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt:
 Retransmission timeout reached on transmission
 OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response)
 -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 Packet timed out after 32000ms with no response
 [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up
 call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical
 packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 ).
 [Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The
 canary is no more.  He has ceased to be!  He's expired and gone to meet his
 maker!  He's a stiff!  Bereft of life, he rests in peace.  His metabolic
 processes are now history!  He's off the twig!  He's kicked the bucket.
  He's shuffled off his mortal coil, run down the curtain, and joined the
 bleeding choir invisible!!  THIS is an EX-CANARY.  (Reducing priority)


 Thanks.

 --
 Date: Thu, 19 Sep 2013 13:14:59 +0500
 From: msalman...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] The call is established but without
 exchanged voice packets

 Choose suitable NAT settings from sip.conf

 turn direct media in sip.conf or per peer off


 On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.comwrote:

 Hello,

 I am trying to make my first call on Asterisk to succeed. I have Asterisk
 1.8.10.1 running on Ubuntu machine.
 The configuration is quite simple just for my first test, Trying to have a
 call between two X-lite sipphone. The subscribers succeeded to register and
 the call is established, but still no voice can be heard, and lead the
 call to be disconnected after! By checking the logs, I can see this

Re: [asterisk-users] The call is established but without exchanged voice packets

2013-09-20 Thread Asghar Mohammad

Hello,
paste you extension context.


On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:

 Hello,

 I have Asterisk 1.8.10.1
 Moving to nat=force_rport,comedia hasn't solved the problem. Still having
 the same error!

 I am not sure if this is related to the problem here, but I was trying to
 test my voicemail and got this error No audio available).
 [Sep 20 14:05:41] WARNING[11424]: app_dial.c:2218 dial_exec_full: Unable
 to create channel of type 'SIP' (cause 20 - Unknown)
 [Sep 20 14:05:54] WARNING[11424]: app.c:855 __ast_play_and_record: No
 audio available on SIP/7001-0001??
 [Sep 20 14:06:13] WARNING[11387]: chan_sip.c:3641 retrans_pkt:
 Retransmission timeout reached on transmission
 ZjJkNTY0YzZjMTcwNzcwYTg0NWRiMjlhYzQ4ZjFkOTc for seqno 2 (Critical Response)
 -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions


 Thanks.

 --
 Date: Fri, 20 Sep 2013 16:05:35 +0200
 From: asghar...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] The call is established but without
 exchanged voice packets

 Hello,
 If Asterisk version is  1.6 use nat=force_rport,comedia


 On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.comwrote:

 Hello,

 I have set the direct media to be off, but still doesn't work. I am not
 sure about NAT configuration!

 SIP.conf, [general] section
 context=internal
 allowguest=no
 allowoverlap=no
 transport=udp
 bindport=5060
 bindaddr=0.0.0.0
 directmedia=no
 srvlookup=no
 disallow=all
 allow=ulaw
 alwaysauthreject=yes
 canreinvite=no
 nat=yes
 session-timers=refuse
 externip=IP
 localnet=172.16.0.255/255.255.255.0

 The error messages

 [Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728
 handle_request_subscribe: Received SIP subscribe for peer without mailbox:
 7002
 [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt:
 Retransmission timeout reached on transmission
 OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response)
 -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 Packet timed out after 32000ms with no response
 [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up
 call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical
 packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 ).
 [Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The
 canary is no more.  He has ceased to be!  He's expired and gone to meet his
 maker!  He's a stiff!  Bereft of life, he rests in peace.  His metabolic
 processes are now history!  He's off the twig!  He's kicked the bucket.
  He's shuffled off his mortal coil, run down the curtain, and joined the
 bleeding choir invisible!!  THIS is an EX-CANARY.  (Reducing priority)


 Thanks.

 --
 Date: Thu, 19 Sep 2013 13:14:59 +0500
 From: msalman...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] The call is established but without
 exchanged voice packets

 Choose suitable NAT settings from sip.conf

 turn direct media in sip.conf or per peer off


 On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.comwrote:

 Hello,

 I am trying to make my first call on Asterisk to succeed. I have Asterisk
 1.8.10.1 running on Ubuntu machine.
 The configuration is quite simple just for my first test, Trying to have a
 call between two X-lite sipphone. The subscribers succeeded to register and
 the call is established, but still no voice can be heard, and lead the
 call to be disconnected after! By checking the logs, I can see this
 chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on
 transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2
 (Critical Response)

 Here's my  simple sip configuration
 [general]
 context=internal
 allowguest=no
 allowoverlap=no
 bindport=5060
 bindaddr=0.0.0.0
 srvlookup=no
 disallow=all
 allow=ulaw
 alwaysauthreject=yes
 canreinvite=no
 nat=yes
 session-timers=refuse
 externip=IP

 [7001]
 type=friend
 host=dynamic
 secret=123
 context=internal

 [7002]
 type=friend
 host=dynamic
 secret=456
 context=internal

 A snoop capture  for my call is uploaded in the following link. I wonder
 if there is any missing configuration or plugin need to be set here!

 http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992
  
 http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992
 Thanks.


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

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users




 --
 Regards

 **
 Muhammad Salman
 ***


 --

Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-19 Thread Asghar Mohammad

remove content of /var/log/asterisk/messages  /var/log/asterisk/messages
run asterisk and post content of /var/log/asterisk/messages to pastebin.


On Thu, Sep 19, 2013 at 9:39 AM, Asmaa Ahmed asabatg...@hotmail.com wrote:

 Hello,

 No, another installation haven't solved the problem!
 It looks more like something related to the configuration in setting the
 running environment!

 Thanks.

 --
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Re: [asterisk-users] sipgate outgoing calls

2013-09-19 Thread Asghar Mohammad

you have insecure=port,invite in sipgate peer?


On Thu, Sep 19, 2013 at 12:26 PM, gpxctawjc...@irational.org wrote:

 On Thu, 19 Sep 2013, Miguel Oyarzo wrote:


 Challenge authentication look good.

 --- SIP read from UDP:217.10.79.23:5060 ---
 SIP/2.0 200 OK

 Are you sure this number format  01179553708 is accepted in that SIP
 trunk?
 Some VOIP providers only accept international format.


 when i use a softphone client to connect directly to sipgate
 i can dial 01179553708 and get through

 --
 __**__**_
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 http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Asghar Mohammad

become root sudo su - or su -l give your password.
if asterisk is already running connect to asterisk -rvvvc otherwisw
asterisk -c.
if you want asterisk run as daemon asterisk and then connect to asterisk
asterisk -rvvvc


On Wed, Sep 18, 2013 at 2:13 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:

 Hello,

 I have started using Asterisk recently on my Ubuntu server. I installed it
 first using apt-get and it worked fine sort of, but still couldn't hear
 voice during the call!
 I read that this problem solved by reinstalling it, so I decided to
 reinstall the latest version from the source as apt-get can give you only
 Asterisk 1.8
 So I have installed Asterisk 11 following the procedure in Asterisk- The
 Definitive Guide, 4th Edition book
 The installation done with no problem, but now I can't even login to CLI,
 it keeps returns Linux prompt!
 ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r
 ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc
 ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv
 ubuntu@vbefe01:/etc/asterisk$

 and if I don't use sudo, it returns a core dump
 $ asterisk -vvc
 Illegal instruction (core dumped)

 I was able to connect before installing the latest version from the
 source? How can I fix that?

 Thanks.

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Asghar Mohammad

SELinux  exists in Ubuntu?



On Wed, Sep 18, 2013 at 2:45 PM, Ishfaq Malik i...@pack-net.co.uk wrote:

 Have you checked your SELinux settings?


 On 18 September 2013 13:13, Asmaa Ahmed asabatg...@hotmail.com wrote:

 Hello,

 I have started using Asterisk recently on my Ubuntu server. I installed
 it first using apt-get and it worked fine sort of, but still couldn't hear
 voice during the call!
 I read that this problem solved by reinstalling it, so I decided to
 reinstall the latest version from the source as apt-get can give you only
 Asterisk 1.8
 So I have installed Asterisk 11 following the procedure in Asterisk- The
 Definitive Guide, 4th Edition book
 The installation done with no problem, but now I can't even login to CLI,
 it keeps returns Linux prompt!
 ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r
 ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc
 ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv
 ubuntu@vbefe01:/etc/asterisk$

 and if I don't use sudo, it returns a core dump
 $ asterisk -vvc
 Illegal instruction (core dumped)

 I was able to connect before installing the latest version from the
 source? How can I fix that?

 Thanks.

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

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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 --

 Ishfaq Malik
 Department: VOIP Support
 Company: Packnet Limited
 t: +44 (0)845 004 4994
 f: +44 (0)161 660 9825
 e: i...@pack-net.co.uk
 w: http://www.pack-net.co.uk

 Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
 37 Ducie Street
 Manchester, M1 2JW
 COMPANY REG NO. 04920552


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Re: [asterisk-users] Can't connect to Asterisk cli

2013-09-18 Thread Asghar Mohammad

i think you messed 2 installs of asterisk.
if you compile asterisk from sources it not insert init script.
you can test installing to /opt.

1. cd to asterisk sources folder
2. make distclean
3. ./configure --prefix=/opt/asterisk
4. make
5. sudo make install
6. /opt/asterisk/sbin/asterisk -c
you can remove this installation by sudo rm -rfv /opt/asterisk
if it work then you should remove every asterisk installation and then
install a fresh copy .(or reinstall OS if you cannot remove)
hope this will help you.



On Wed, Sep 18, 2013 at 2:57 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:

 It looks this is because Asterisk isn't started
 when tried to start it, I got a core dump!
 $ /etc/init.d/asterisk start
  * Starting Asterisk PBX: asterisk
 Illegal instruction (core dumped)

 --
 From: asabatg...@hotmail.com
 To: asterisk-users@lists.digium.com
 Subject: Can't connect to Asterisk cli
 Date: Wed, 18 Sep 2013 14:13:16 +0200

 Hello,

 I have started using Asterisk recently on my Ubuntu server. I installed it
 first using apt-get and it worked fine sort of, but still couldn't hear
 voice during the call!
 I read that this problem solved by reinstalling it, so I decided to
 reinstall the latest version from the source as apt-get can give you only
 Asterisk 1.8
 So I have installed Asterisk 11 following the procedure in Asterisk- The
 Definitive Guide, 4th Edition book
 The installation done with no problem, but now I can't even login to CLI,
 it keeps returns Linux prompt!
 ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r
 ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc
 ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv
 ubuntu@vbefe01:/etc/asterisk$

 and if I don't use sudo, it returns a core dump
 $ asterisk -vvc
 Illegal instruction (core dumped)

 I was able to connect before installing the latest version from the
 source? How can I fix that?

 Thanks.

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

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] Asterisk-1.8.23.1 mysql cdr

2013-09-17 Thread Asghar Mohammad

Hi list,
Reply to my own question
http://lists.digium.com/pipermail/asterisk-users/2013-September/280541.html

I come up with a patch that enable timezone support.
add new configuration cdrzone option in cdr_mysql.conf.
in cdr_mysql.conf add cdrzone= any valid timezone(consult
/usr/share/zoneifo) disable all existing options usegmtime etc.
added new cli option cdr mysql cdrzone. it will show you selected timezone.
patch can be download from
http://www.world-call-trade.com/asterisk/cdr_mysql_cdrzone.patch
please report back here.

BEST REGARDS
Asghar Mohammad
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[asterisk-users] Asterisk-1.8.23.1 mysql cdr

2013-09-14 Thread Asghar Mohammad

Hi list,
I am using Asterisk1.6.2 form a long time and upgarding to
Asterisk-1.8.23.1.
I am using mysql backend for cdr.
in asterisk-1.6.2 i have usegmtime=yes and it works as expected insert cdr
date in GMT0.
now i tested Asterisk-1.8.23.1 and asterisk-11.5 with same results no
matter what i configure in cdr_mysql.conf timezone=UTC usegmtime=yes cdr
always inserted in local time.

I dig into code of cdr_mysql.c and find a variable cdrzone when i set
cdrzone in configuration and load module with debug set to 1 it print on
console Local time zone set to whatever i have in configuration i tried
cdrzone=GMT, cdrzone=UTC, cdrzone=yes and many combinations with
timezone=UTC and without timezone=UTS but cdr is alway in my local timezone
GMT +2.

in further investigation i have seen there is no timezone conversation.
from asterisk1.8.231 mysql_cdr.c

  if (!strcmp(entry-name, calldate)) {
/*!\note
 * For some dumb reason, calldate used to
be formulated using
 * the datetime the record was posted,
rather than the start
 * time of the call.  If someone really
wants the old compatible
 * behavior, it's provided here.
 */
if (calldate_compat) {
struct timeval tv = ast_tvnow();
struct ast_tm tm;
char timestr[128];
ast_localtime(tv, tm,
ast_str_strlen(cdrzone) ? ast_str_buffer(cdrzone) : NULL);
ast_strftime(timestr,
sizeof(timestr), %Y-%m-%d %T, tm);
ast_cdr_setvar(cdr, calldate,
timestr, 0);
cdrname = calldate;
} else {
cdrname = start;
}
} else {
cdrname = entry-cdrname;
}


from addons  1.6.2.4 mysql_cdr.c

 if (calldate_compat) {
struct timeval tv = ast_tvnow();
struct ast_tm tm;
char timestr[128];
ast_localtime(tv, tm, NULL);
ast_strftime(timestr,
sizeof(timestr), %Y-%m-%d %T, tm);
ast_cdr_setvar(cdr, calldate,
timestr, 0);
cdrname = calldate;
} else if (usegmtime) {
struct ast_tm tm;
char timestr[128];
ast_localtime(cdr-start, tm,
GMT);
ast_strftime(timestr,
sizeof(timestr), DATE_FORMAT, tm);
ast_cdr_setvar(cdr, calldate,
timestr, 0);
cdrname = calldate;
} else {
cdrname = start;
}
} else {
cdrname = entry-cdrname;

please note else if(usegmtime)
the codes are removed from latest asterisk versions.
i am not c programmer anybody can help me solve this issue?

Thanks in advance.
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Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes

2013-09-10 Thread Asghar Mohammad

hi,
it seems your vpn connection drop.
is you vpn on WiFi of any other high latency network?


On Tue, Sep 10, 2013 at 1:05 PM, Administrator TOOTAI ad...@tootai.netwrote:

 Hi all,

 I face the subject strange behavior: calls arre dropped after 15 minutes
 on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk
 through OpenVPN seems to have the problem.

 From CDR, I see for 3 calls from this morning I'm aware of, that asterisk
 hangup after respectively 899s 894s 898s

 In logs I see

 WARNING[8213] chan_sip.c: Retransmission timeout reached on transmission
 522eec628683-uy8xshd6wc21 for seqno 102 (Critical Request) -- See
 https://wiki.a
 Packet timed out after 6401ms with no response (or 6399ms or 6401ms)

 Qualify freq being 6 ms for the peers.

 For the SIP peers I see

   Sess-Timers  : Accept
   Sess-Refresh : uas
   Sess-Expires : 1800 secs
   Min-Sess : 90 secs

 I tried to use originate for session-timer in global SIP conf, no changes.

 Any hint about this matter would be appreciate.

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Re: [asterisk-users] No remote address on RTP instance - On Ringing

2013-09-10 Thread Asghar Mohammad

i have used a2billing some time ago maybe there is somthing new .
you can try shoot up loglevel to 4 and see the verbose of agi that may give
you some hint.




On Tue, Sep 10, 2013 at 7:34 PM, jg webaccou...@jgoettgens.de wrote:


  Maybe the ringtone from downstream is not
 reaching asterisk, and thus a2billing is appending the `m` to the dial
 command?

 With digital systems (starting with ISDN, or so), ringing is signaled,
 or indicated. The ringtone is produced locally, either by the PBX or by the
 SIP phone itself. Since you do get the invitation, everything is fine.

 If you really can't remove the m, you could still use an audio file with
 a funny ringtone and stuff this into an moh class. Dirty, but it will work.

 jg

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Re: [asterisk-users] Cisco SPA303 won't ring for more than 60 seconds

2013-08-21 Thread Asghar Mohammad

sip set debug on and see trace of upload on pastebin.


On Wed, Aug 21, 2013 at 8:25 PM, jg webaccou...@jgoettgens.de wrote:

 At first I also thought this might be a phone setting. But then I found
 the same 60s to be true for a variety of SIP phones (Snom, Cisco, ...),
 despite the 300s timeout value in the Dial cmd. So it is likely to be
 Asterisk. The Asterisk Admin Guide says that the default value is 136
 years, so there must be something that sets this timeout value.

 jg

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Re: [asterisk-users] Am I being hacked?

2013-08-19 Thread Asghar Mohammad

he,
some bad boys trying to guess configured extensions.
in sip config in general set alwaysauthreject = yes .
in cli sip set debug on and watch ip and block in firewall, iptables.


On Mon, Aug 19, 2013 at 7:50 PM, Ira i...@extrasensory.com wrote:

  Hello Steve,

 Sunday, August 18, 2013, 3:35:54 PM, you wrote:

  On Sun, 18 Aug 2013, Ira wrote:

  [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c:
 Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx
 ;tag=2762c06e
 
  I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my
 own
  IP.  How do I figure out where this attempt is coming from so I can
  block it.

  Any chance '390' is a legitimate (but mis-configured or obsolete) device
  on your network?

  Is xx.xx.xxx.xxx a private or public address?

  Can you 'wireshark' some packets and see if the OUI matches one of your
  endpoints?

 390 is not, nor has it ever been an extension on my box. I've gotten the
 same message for numerous extensions, sometimes 100-200 inclusive, usually
 multiple times as if they are trying multiple passwords.  I'm sure that no
 one will ever guess an extension or password on my box that way so I'm not
 worried, I've blocked most of the IPs that my box doesn't use and it's been
 a long time since I've seen any outside attempts to register. But in the
 recent past I've been seeing these where I've no clue what IP to block as
 the entries, sip:3...@xx.xx.xxx.xxx, always contains an invalid extension
 and my cable modem's IP address.

 xx.xx.xxx.xxx is my public I.P.

 I searched Google and found no mention of my specific error.

 -- Ira

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Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

2013-08-18 Thread Asghar Mohammad

just remove username.
type peer authenticate by ip


On Sun, Aug 18, 2013 at 7:01 PM, Andrew Colin and...@vsave.co.za wrote:

  change server two to host = dynamic

 then add register = on server 1

 On 8/18/2013 6:29 PM, Gopalakrishnan N wrote:

 Even I tried the type as friend.. but no use...


 On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi,

  Am making a simple SIP trunk between two Asterisk server,

  Server 1
 sip.conf
  [usman02]
 type=peer
 username=usman02
 secret=usman02
 host=10.30.2.58
 context=man02-trunk
 port=5060
 qualify=yes
 disallow=all
 ;allow=g729
 allow=g729
 ;allow=alaw
 nat=force_rport,comedia
 dtmfmode=rfc2833
 relaxdtmf=yes
 insecure=invite,port

  extensions.conf
  [man02-trunk]
 exten = _1X.,1,Dial(SIP/usman02/${EXTEN})
 exten = _1X.,n,Hangup


  Server2
 sip.conf
  [usman02]
 type=peer
 username=usman02
 secret=usman02
 host=10.10.10.81
 context=us02-trunk-inbound
 port=5060
 qualify=yes
 disallow=all
 allow=g729
 ;allow=ulaw
 ;allow=alaw
 nat=force_rport,comedia
 dtmfmode=rfc2833
 relaxdtmf=yes
  insecure=port,invite

  extensions.conf
  [us02-trunk-inbound]
 exten = _X.,Dial(SIP/${EXTEN},60)


  Now when I dial from server1, in the server 2 am getting the error as,
 [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth:
 username mismatch, have 2001, digest has usman02

  things are fine.. but I dont know where the mistake is...!

  Can you some one advise me... !

  Thanks.




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Re: [asterisk-users] G729 Passthrough How To

2013-08-14 Thread Asghar Mohammad

As my understanding Asterisk always pass-thu g729 if both ends have this
codec.
But if you answer the call or play some audio before dialing to end point
then asterisk stay between both legs.
In case of VM. you should install g729 if your prompts are in g729 format.
As a2billing play voice prompts you cannot pass-thu transparently.
I think the load on you server is not for transcoding  but PHP scripts.
I was in this situation and reduce the upto 80% by removing A2B.


On Wed, Aug 14, 2013 at 4:44 PM, Nick Khamis sym...@gmail.com wrote:

 Hey!!! Eric thank you so much for your response. Could you guys please
 direct us in achieving as much as possible. For example:
 * What linux command can we use to convert all recording to G729
 * Which files do we need to convert and there locations
 * For *testing* how do we make sure Asterisk NEVER EVER transcodes.

 Do we still need the G729 codec installed on the asterisk machine if
 we manage to implement pass-through that would suffice our needs.

 Kind Regards,

 Nick.

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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad

hi,
paste server a trunk also, if you want register why you are not using
host=dynamic?
both servers are on 10.10.10.0 ? if no then check your deny permit seting.


On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other IP not
 registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue on the network side...

 Problem is there is no packet loss.. with mtr it is fine, tracepath is
 fine, ping is fine... :(


 On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Am using Asterisk 11.2 in one location and 11.1 in another location.

 when I trunk between two servers, the status is unreachable.

 But with different server with 11.2 and 11.2 it works fine.

 I tried both IAX and SIP.

 the trunk in sip.conf what i have is,
 [serverb]
 type=friend
 username=serverb
 secret=serverb
 host=10.10.10.5
 port=5060
 context=default
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=3000
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.10.10.5/255.255.255.0

 Is there any issue with 11.1?



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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad

1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b and
10.10.10.0 on a.
2. use host=dynamic type=friend on  side A and host=ip type=peer on side B.
3. general section in sip.conf of side B register with server A.

please see comments in sip.conf
;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
registering
; as any IP address used for staticly
defined
; hosts.  This helps avoid the configuration
; error of allowing your users to register
at
; the same address as a SIP provider.



On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.30.2.5/255.255.255.0

 If i use host=dynamic, it wont communicate each other and will result to
 unmonitored


 and the IP segment is two different segment. where am able to ping each
 other.



 On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad asghar...@gmail.comwrote:

 hi,
 paste server a trunk also, if you want register why you are not using
 host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other IP not
 registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue on the network side...

 Problem is there is no packet loss.. with mtr it is fine, tracepath is
 fine, ping is fine... :(


 On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Am using Asterisk 11.2 in one location and 11.1 in another location.

 when I trunk between two servers, the status is unreachable.

 But with different server with 11.2 and 11.2 it works fine.

 I tried both IAX and SIP.

 the trunk in sip.conf what i have is,
 [serverb]
 type=friend
 username=serverb
 secret=serverb
 host=10.10.10.5
 port=5060
 context=default
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=3000
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.10.10.5/255.255.255.0

 Is there any issue with 11.1?



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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad

yes you can. just create trunks on both side with static ip and in dial use
trunk name.
exten = _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =
_X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
make a call from a to b and one from b to and post cli log here or upload
anyware else.


On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 can't we use without register command both way as peer to peer?


 On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad asghar...@gmail.comwrote:

 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b and
 10.10.10.0 on a.
 2. use host=dynamic type=friend on  side A and host=ip type=peer on side
 B.
 3. general section in sip.conf of side B register with server A.

 please see comments in sip.conf
 ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
 registering
 ; as any IP address used for staticly
 defined
 ; hosts.  This helps avoid the
 configuration
 ; error of allowing your users to
 register at
 ; the same address as a SIP provider.



 On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.30.2.5/255.255.255.0

 If i use host=dynamic, it wont communicate each other and will result to
 unmonitored


 and the IP segment is two different segment. where am able to ping each
 other.



 On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad asghar...@gmail.comwrote:

 hi,
 paste server a trunk also, if you want register why you are not using
 host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit
 seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other IP not
 registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue on the network side...

 Problem is there is no packet loss.. with mtr it is fine, tracepath is
 fine, ping is fine... :(


 On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Am using Asterisk 11.2 in one location and 11.1 in another location.

 when I trunk between two servers, the status is unreachable.

 But with different server with 11.2 and 11.2 it works fine.

 I tried both IAX and SIP.

 the trunk in sip.conf what i have is,
 [serverb]
 type=friend
 username=serverb
 secret=serverb
 host=10.10.10.5
 port=5060
 context=default
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=3000
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.10.10.5/255.255.255.0

 Is there any issue with 11.1?



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Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad

*1st Location*
[manila]
type=peer
username=indman01
secret=indman01
host=10.30.2.5 -- ip of 2nd location
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw

1st location dialplan
exten = _2XXX,1,Dial(SIP/manila/${EXTEN} http://10.30.2.5/$%7BEXTEN%7D)
exten = _2XXX,n,Hangup

*2nd Location*
[india]
type=friend
username=manind01
secret=manind01
host=dynamic
port=5060
context=10.20.111.48 - ip of 1st location
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw

2st location dialplan
exten = _2XXX,1,Dial(SIP/india/${EXTEN} http://10.30.2.5/$%7BEXTEN%7D)
exten = _2XXX,n,Hangup

then you should handle the call when it arrive in any server
let me know if it work.


On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 I tried creating two trunks with following,
 *1st Location*
 [10.30.2.5]
 type=friend
 username=indman01
 secret=indman01
 host=dynamic
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw

 *2nd Location*
 [10.20.111.48]
 type=friend
 username=manind01
 secret=manind01
 host=dynamic
 port=5060
 context=india
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw

 My dialplan is like this
 exten = _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}http://10.30.2.5/$%7BEXTEN%7D
 )
 exten = _2XXX,n,Hangup

 And the output I get is
  Executing [2001@Test:1] Dial(SIP/3081-27d2, SIP/10.30.2.5/2001)
 in new stack
 [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
 Subscriber absent)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing [2001@Test:2] Hangup(SIP/3081-27d2, ) in new
 stack
   == Spawn extension (Test, 2001, 2) exited non-zero on 'SIP/3081-27d2'

 Actually the trunk which i mentioned in my first email, it was working...
 and from today it is not

 Still breaking... what could be the reason... !



 On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad asghar...@gmail.comwrote:

 yes you can. just create trunks on both side with static ip and in dial
 use trunk name.
 exten = _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =
 _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
 make a call from a to b and one from b to and post cli log here or upload
 anyware else.


 On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 can't we use without register command both way as peer to peer?


 On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad asghar...@gmail.comwrote:

 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
 and 10.10.10.0 on a.
 2. use host=dynamic type=friend on  side A and host=ip type=peer on
 side B.
 3. general section in sip.conf of side B register with server A.

 please see comments in sip.conf
 ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
 registering
 ; as any IP address used for staticly
 defined
 ; hosts.  This helps avoid the
 configuration
 ; error of allowing your users to
 register at
 ; the same address as a SIP provider.



 On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.30.2.5/255.255.255.0

 If i use host=dynamic, it wont communicate each other and will result
 to unmonitored


 and the IP segment is two different segment. where am able to ping
 each other.



 On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad 
 asghar...@gmail.comwrote:

 hi,
 paste server a trunk also, if you want register why you are not using
 host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit
 seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other IP
 not registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue on the network side...

 Problem is there is no packet loss.. with mtr it is fine, tracepath
 is fine, ping is fine... :(


 On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Am using Asterisk 11.2 in one location and 11.1 in another
 location.

 when I trunk between two servers

Re: [asterisk-users] Asterisk trunking between two location

2013-07-02 Thread Asghar Mohammad

make a call and post cli log


On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 still the peer shows unreachable let me restart and give a try...


 On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad asghar...@gmail.comwrote:

 *1st Location*
 [manila]
 type=peer
 username=indman01
 secret=indman01
 host=10.30.2.5 -- ip of 2nd location
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw

 1st location dialplan
 exten = _2XXX,1,Dial(SIP/manila/${EXTEN} http://10.30.2.5/$%7BEXTEN%7D
 )
 exten = _2XXX,n,Hangup

 *2nd Location*
 [india]
 type=friend
 username=manind01
 secret=manind01
 host=dynamic
 port=5060
 context=10.20.111.48 - ip of 1st location
  insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw

 2st location dialplan
 exten = _2XXX,1,Dial(SIP/india/${EXTEN} http://10.30.2.5/$%7BEXTEN%7D)
 exten = _2XXX,n,Hangup

 then you should handle the call when it arrive in any server
 let me know if it work.


 On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 I tried creating two trunks with following,
 *1st Location*
 [10.30.2.5]
 type=friend
 username=indman01
 secret=indman01
 host=dynamic
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw

 *2nd Location*
 [10.20.111.48]
 type=friend
 username=manind01
 secret=manind01
 host=dynamic
 port=5060
 context=india
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw

 My dialplan is like this
 exten = _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}http://10.30.2.5/$%7BEXTEN%7D
 )
 exten = _2XXX,n,Hangup

 And the output I get is
  Executing [2001@Test:1] Dial(SIP/3081-27d2, SIP/10.30.2.5/2001)
 in new stack
 [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
 Subscriber absent)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing [2001@Test:2] Hangup(SIP/3081-27d2, ) in new
 stack
   == Spawn extension (Test, 2001, 2) exited non-zero on
 'SIP/3081-27d2'

 Actually the trunk which i mentioned in my first email, it was
 working... and from today it is not

 Still breaking... what could be the reason... !



 On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad asghar...@gmail.comwrote:

 yes you can. just create trunks on both side with static ip and in dial
 use trunk name.
 exten = _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =
 _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
 make a call from a to b and one from b to and post cli log here or
 upload anyware else.


 On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 can't we use without register command both way as peer to peer?


 On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad 
 asghar...@gmail.comwrote:

 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
 and 10.10.10.0 on a.
 2. use host=dynamic type=friend on  side A and host=ip type=peer on
 side B.
 3. general section in sip.conf of side B register with server A.

 please see comments in sip.conf
 ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
 registering
 ; as any IP address used for staticly
 defined
 ; hosts.  This helps avoid the
 configuration
 ; error of allowing your users to
 register at
 ; the same address as a SIP provider.



 On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.30.2.5/255.255.255.0

 If i use host=dynamic, it wont communicate each other and will
 result to unmonitored


 and the IP segment is two different segment. where am able to ping
 each other.



 On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad asghar...@gmail.com
  wrote:

 hi,
 paste server a trunk also, if you want register why you are not
 using host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit
 seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other IP
 not registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue

Re: [asterisk-users] how to send dtmf after pause ?

2013-06-07 Thread Asghar Mohammad

hi,
you can add more w (ww1234#) for more delay.

On Fri, Jun 7, 2013 at 7:17 PM, Yves A. yves...@gmx.de wrote:

This would be possible with an agi...
the agi can wait for silence or 10 seconds, as u like and then play the
dtmf tones and bridge the call to your extension afterwards.

yves

Am 07.06.2013 17:51, schrieb Sean Darcy:

I'm trying to call a conference service, wait 10 seconds, then send the
passcode.

I've tried ww:

Dial(SIP/18005551212ww12345#@s**ip.com http://sip.com,60,r)

The sip channel didn't like that. Added 'p' , still no help.

I tried D:

Dial(SIP/18005551...@sip.com,**60,rD(12345#)

The dtmf is sent too soon. I tried inserting 'ww' but that was just sent.

I tried G:

exten = 234.1.Dial(SIP/18005551212@**sip.com 18005551...@sip.com
,60,rG(next))
same=n(next),Wait(10)
same=n,SendDTMF(12345#)

but that didn't work at all,

This is a common use case. There must be some simple answer I'm missing.

Thanks for any help.

sean