Re: [asterisk-users] Notifying missed calls

[Carlos Rojas]

g: *When the called party hangs up*, continue to execute commands in the
current context at the next priority

On Wed, Nov 3, 2021 at 4:39 PM Luca Bertoncello 
wrote:

> Am 03.11.2021 um 21:34 schrieb Antony Stone:
> > On Wednesday 03 November 2021 at 21:29:46, Luca Bertoncello wrote:
> >
> >> I tried so:
> >>
> >> exten => h,n(hang),Gosub(noanswer,s,1)
> >
> > The n there should be 1, surely?
>
> Ach, you're right!
>
> Now it works!
>
> Thanks a lot
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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Re: [asterisk-users] Community forum ?

[Carlos Rojas]

Do you know what is going to happen with the DCAP certificates?

Are they going to be valid?



On Thu, Aug 30, 2018 at 11:29 AM, Matthew Jordan  wrote:

>
>
> On Thu, Aug 30, 2018 at 3:25 PM John Covici  wrote:
>
>> Is Sangoma taking over Digium?  Pretty soon there won't be anything
>> open source around in this field at all.
>>
>>
> Sangoma acquired Digium.
>
> How this impacts Asterisk is answered by the community FAQ:
>
> https://wiki.asterisk.org/wiki/display/AST/Sangoma+and+
> Digium+Join+Together+FAQ
>
> tl;dr: it doesn't.
>
>
>
>
>> On Thu, 30 Aug 2018 11:14:33 -0400,
>> Carlos Rojas wrote:
>> >
>> > [1  ]
>> > [1.1  ]
>> > [1.2  ]
>> > Is the list going to be the same after sangoma take over digium?
>> >
>> > On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp  wrote:
>> >
>> >  On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote:
>> >  > I see a lot of tag lines on posts for the Asterisk Community Forum.
>> Is
>> >  > that forum supposed to supersede this mailing list ?
>> >
>> >  Both remain available but the community forum seems to be more active,
>> and it is easier to search and find things.
>> >
>> >  --
>> >  Joshua Colp
>> >  Digium, Inc. | Senior Software Developer
>> >  445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> >  Check us out at: www.digium.com & www.asterisk.org
>> >
>> >  --
>> >  _
>> >  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >
>> >  Astricon is coming up October 9-11!  Signup is available at:
>> https://www.asterisk.org/community/astricon-user-conference
>> >
>> >  Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>> >
>> >  New to Asterisk? Start here:
>> >https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>> >
>> >  asterisk-users mailing list
>> >  To UNSUBSCRIBE or update options visit:
>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> > [2  ]
>> > --
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>> > Astricon is coming up October 9-11!  Signup is available at:
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>> >
>> > Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>> >
>> > New to Asterisk? Start here:
>> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>>  John Covici wb2una
>>  cov...@ccs.covici.com
>>
>> --
>> _
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>>
>> Astricon is coming up October 9-11!  Signup is available at:
>> https://www.asterisk.org/community/astricon-user-conference
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> Matthew Jordan
> Digium, Inc. | CTO
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
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>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Community forum ?

[Carlos Rojas]

Yes it is.

https://www.sangoma.com/press-releases/sangoma-announces-definitive-agreement-to-acquire-digium-inc/


https://wiki.freepbx.org/display/FOP/Sangoma+and+Digium+Join+Together+FAQ




On Thu, Aug 30, 2018 at 11:25 AM, John Covici  wrote:

> Is Sangoma taking over Digium?  Pretty soon there won't be anything
> open source around in this field at all.
>
> On Thu, 30 Aug 2018 11:14:33 -0400,
> Carlos Rojas wrote:
> >
> > [1  ]
> > [1.1  ]
> > [1.2  ]
> > Is the list going to be the same after sangoma take over digium?
> >
> > On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp  wrote:
> >
> >  On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote:
> >  > I see a lot of tag lines on posts for the Asterisk Community Forum.
> Is
> >  > that forum supposed to supersede this mailing list ?
> >
> >  Both remain available but the community forum seems to be more active,
> and it is easier to search and find things.
> >
> >  --
> >  Joshua Colp
> >  Digium, Inc. | Senior Software Developer
> >  445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> >  Check us out at: www.digium.com & www.asterisk.org
> >
> >  --
> >  _
> >  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> >  Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
> >
> >  Check out the new Asterisk community forum at:
> https://community.asterisk.org/
> >
> >  New to Asterisk? Start here:
> >https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> >  asterisk-users mailing list
> >  To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > [2  ]
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
> >
> > Check out the new Asterisk community forum at:
> https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>  John Covici wb2una
>  cov...@ccs.covici.com
>
> --
> _
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>
> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Community forum ?

[Carlos Rojas]

Is the list going to be the same after sangoma take over digium?

On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp  wrote:

> On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote:
> > I see a lot of tag lines on posts for the Asterisk Community Forum. Is
> > that forum supposed to supersede this mailing list ?
>
> Both remain available but the community forum seems to be more active, and
> it is easier to search and find things.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Community forum ?

[Carlos Rojas]

I don't think so.

On Thu, Aug 30, 2018 at 11:05 AM, sean darcy  wrote:

> I see a lot of tag lines on posts for the Asterisk Community Forum. Is
> that forum supposed to supersede this mailing list ?
>
> sean
>
>
> --
> _
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>
> Astricon is coming up October 9-11!  Signup is available at:
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
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Re: [asterisk-users] getting invites to rtp ports ??

[Carlos Rojas]

Hi

Probably somebody is trying to hack your system, you should block that ip
on your firewall.

Regards

On Wed, Aug 29, 2018 at 9:34 AM, sean darcy  wrote:

> I'm getting invites to very high ports every 30 seconds from a particular
> ip address:
>
> Retransmitting #10 (NAT) to 5.199.133.128:52734:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 0.0.0.0:52734;branch=z9hG4bK12
> 07255353;received=5.199.133.128;rport=52734
> From: ;tag=1872048972
> To: ;tag=as3a52e748
> Call-ID: 1504207870-295758084-609228182
> CSeq: 1 INVITE
> ...
> WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
> 1504207870-295758084-609228182...
>
> I thought invites had to go to port 5060 or so. I don't understand why
> somebody (let's assume a bad guy) is trying ports above 5.
>
> sean
>
>
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Re: [asterisk-users] Pass through registration / proxy

[Carlos Rojas]

Hi

You could use kamailio +asterisk

On Tue, Apr 10, 2018, 9:25 PM Telium Technical Support 
wrote:

> I need to create a SIP proxy to be placed in front of a legacy PBX.  When
> a phone registers with the proxy, I would like Asterisk to register with
> the PBX behind it.  (To tell the PBX to send calls to the proxy and then to
> the SIP phone).
>
>
>
> Can I use Asterisk to create a proxy like this?  Is there a way to cause
> the Asterisk to register with another host when it receives a successfully
> registration?
>
>
>
> Thanks!
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Re: [asterisk-users] Using g729 now that patents have expired

[Carlos Rojas]

Hi

You can uses:

http://asterisk.hosting.lv/



On Tue, Feb 7, 2017 at 4:47 PM, Steve Edwards 
wrote:

> Now that the g729 patents have expired, how do we use g729 in Asterisk?
>
> Will Digium be releasing a g729 codec for 'free' use or do we download the
> 'free' codec off the Internet now that we can use it without moral or legal
> restrictions?
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
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Re: [asterisk-users] sip show [general]?

[Carlos Rojas]

Hi

You can do

sip show settings


On Jan 11, 2017 5:32 AM, "Thufir Hawat"  wrote:

> I appreciate that the console lets you see the details for a peer with
> "sip show peer foo".  Certainly, I can look in sip.conf to see the
> [general] context, but can I output those settings, and only those
> settings, to the console?
>
>
>
> thanks,
>
> Thufir
>
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Re: [asterisk-users] implementing call center using asterisk

[Carlos Rojas]

Hi

You can use, gnudialer, vicidial, goautodial.




On Wed, Jun 22, 2016 at 12:47 PM, Goke Aruna  wrote:

> hello all,
> I am looking for an implementation of a 10 man call center.  low cost
> license or GPL will be preferred.
> I will be glad for your help.
> Regards
>
> --
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Re: [asterisk-users] SPA112 flapping

[Carlos Rojas]

Hi

It sounds like a keep alive issue

On Sun, Jun 19, 2016, 4:39 PM Gergo Csibra  wrote:

> Friday, June 17, 2016, 11:56:34 PM, Mike wrote:
>
> > I've got a device that seems to become unreachable for about 2 minutes,
> every
> > hour.  From what I can tell, it isn't due to network or server issues.
> Any
> > ideas?
>
> The default registration time in spa112 is 1 hour. If registering is
> slow in your infrastructure, this can be the reason.
>
> --
> Best regards,
>  Gergomailto:csi...@gmail.com
>
>
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Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

[Carlos Rojas]

I have tried with xen and kvm both are working fine.

On Wed, Apr 6, 2016 at 3:44 PM, Loic Chabert  wrote:

> Hello,
>
> Work well with kvm and centos 7.
> Some ajustements has to be made with systemd.
>
> I'm using it in production since 1.5 year now, no issue to report.
>
> Regards.
> Le 6 avr. 2016 21:13, "Yves"  a écrit :
>
>
> Le 06/04/2016 18:12, Markos Vakondios a écrit :
>
> Good evening,
>
> My English is limited but if I can help.
>
> We install Asterisk Version 13.1 on VmWare with Debian 8.2, no
> problem since June 2015, currently I have tested on Unbutu 14.04 but problem
> with network-manager (problem of stability with Asterisk 1.8.32 and
> difficulty with routing network-manager).
>
> I also installed Asterisk on KVM (Debian 8.2) no problem (but not test
> with dahdi) without particular problem.
>
> here is my little opinion
>
> Hello everyone
>
> Proxmox and KVM on Ubuntu
>
> On Wednesday, 6 April 2016, Ryan, Travis < 
> ry...@oscarwinski.com> wrote:
>
>> What is the best virtual server tech (and most stable, etc) to use for a
>> asterisk virtual hosting environment?
>>
>>
>>
>> I have a client that wants to do virtual hosting of Asterisk (only SIP or
>> IAX, no PRI, etc) and I’m wondering if Xen or something else would be best?
>> We’d like to stay away from the costs of VMWare if possible.
>>
>>
>>
>> Thanks!
>>
>>
>>
>> Travis
>>
>>
>>
>
>
>
>
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Re: [asterisk-users] PRI error "ROSE REJECT"

[Carlos Rojas]

Hi

Did you activate the pri debug on the cli asterisk?

On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez 
wrote:

> We've been having some problems with an E1 PRI line for a few days.  We
> get the following errors:
>
> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT:
> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2INVOKE ID:
> 316
> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2PROBLEM:
> Invoke: Unrecognized Operation
>
> The telephone company says that everything is fine on their side,
> obviously.  The problems started a few days ago when a user reported that
> incoming calls get dropped when you try to dial a particular extension from
> the main IVR.  We are using Asterisk 1.8.15-cert2 on a CentOS 6.7 server,
> DAHDI 2.6.1 and libpri 1.4.  Any recommendations?
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> dCAP #1349
> +52 (55)9116-91161
>
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Re: [asterisk-users] FAX Detection.

[Carlos Rojas]

Hi

I have used sangoma cards, but I know that openvox, is shipper than Sangoma.

On Wed, Feb 24, 2016 at 1:10 PM, Aziz TestAccount 
wrote:

> Hi All,
>
> I'm looking for a PSTN Card that I can use with my Asterisk Server to
> achieve the following goal :
>
> 1. Detect FAX signal and route it to a specific extension.
> 2. Detect an incoming call from the same PSTN line and route it to IVR.
>
> Do openvox FXO/FXS cards support this feature ? Is there any other brand
> that can be used with Asterisk and that is supporting this ?
>
> Thanks in advance.
>
>
>
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Re: [asterisk-users] Looking for Asterisk Consultants & Experts

[Carlos Rojas]

Hi I am Carlos Rojas

I am asterisk dCAP, 2171

What do you need?



On Wed, Sep 2, 2015 at 7:40 AM, Shahid H <shah...@gmail.com> wrote:

> Hello,
>
> Can someone recommend me where is best place to find Asterisk
> Expert/Consultant for freelance work?
>
> If you are interested to work as a freelancer, you can email me directly.
>
> Thanks
>
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Re: [asterisk-users] Grandstream GXP2140

[Carlos Rojas]

Hi

If you are going to use only a phone, it's fine, but if you are going to
install a lot of grandstream's phones, probably you network traffic is
going to increase a lot.

On Wed, Apr 15, 2015 at 3:12 PM, dsi...@hcmr.gr wrote:

 I'm working with GXP2130.
 About 12 phone on gigabit with PC after phone.
 With Vlans on CISCO switch is stable and not so difficult.
 This configuration running without problems since July 2013.



 Quoting jg webaccounts...@jgoettgens.de:

  I have a customer looking to deploy about 20 Grandstream GXP2140 phones.
 Normally they would deploy Yealink brand phones but they want a phone with
 gigabit pass through and the Yealinks with gigabit are too expensive for
 their budget.


 Does anyone on the list have experience with the GXP2140? Is it a
 reliable phone? Does anyone have recommendations for other phones with
 gigabit pass through?


  I'd be generally careful with the second ethernet connection. One
 should look at the chipset of the phone. I had pretty bad experiences with
 somewhat older TI based phones, regardless of the manufacturer. The
 problems became apparent in mixed environments, where some connections were
 gigabit and others not. It can be a nightmare, if you have to offer support.

 The best bet is to buy one, and check the performance of the connections.
 I use some GrandStream products myself and the product quality is now much
 better compared to a couple of years ago.

 jg

 --
 This message has been scanned for viruses and
 dangerous content by MailScanner, and is
 believed to be clean.





 D. Sidirokastritis
 NOC HCMR-Crete
 tel. 2810-337709

 
 Hellenic Center for Marine Research
 This message was sent using IMP, the Internet Messaging Program.



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Re: [asterisk-users] Gateway Eurotech

[Carlos Rojas]

I Ricky

I have worked with this gateway few years ago, it's good product, they have
gateways with PRI connectors and SIP.

The quality is good, and it woks good with asterisk or regular PBXs.

On Thu, Mar 26, 2015 at 11:16 PM, ricky gutierrez xserverli...@gmail.com
wrote:

 Hi, I know there are people with much experience in asterisk, and I
 want to ask if anyone had experiance with this gw

 http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/

 I'm having trouble getting connect with asterisk

 anyone has any production?

 regardss

 --
 rickygm

 http://gnuforever.homelinux.com

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Re: [asterisk-users] Popup URL for outgoing calls.

[Carlos Rojas]

You can use vtiger or sugar

Both are working with asterisk.




On Fri, Jun 27, 2014 at 9:04 PM, Prakash N prakas...@tevatel.com wrote:

 What CRM your going to use?

 With regards

 N.Prakash From: Rusty Newton
 Sent: ‎28-‎06-‎2014 01:01 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Popup URL for outgoing calls.
 On Sat, Jun 21, 2014 at 5:57 AM, Inventions resea...@businesstz.com
 wrote:
  Can anyone tell me how to implement a popup URL native asterisk when
  making outbound call?
 
  For example, a user (A Part) dial from a softphone number 07112233, when
 a
  call is received (or even before) by B-Part, a CRM pops up with
  information for user 07112233 on A-Part computer. More less like incoming
  url popup on a queue.

 No one is going to do the work for you. You'll have to do the
 research. A good place to start is probably the sections in the
 Asterisk Definitive Guide on Asterisk Gateway Interface and Asterisk
 Manager Interface

 http://shop.oreilly.com/product/0636920025894.do

 --
 Rusty Newton
 Digium, Inc. | Community Support Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 direct: +1 256 428 6200

 Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] SIP Softphone

[Carlos Rojas]

Zoiper gsm
-Original Message-
From: Mark Robinson vsysnetw...@gmail.com
Sender: asterisk-users-bounces@lists.digium.comDate: Sun, 8 Jun 2014 17:01:54 
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Softphone

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Re: [asterisk-users] Verbose only one context

[Carlos Rojas]

You can do this

 sip set debug ip x.x.x.x


On Wed, Mar 26, 2014 at 11:28 AM, Rafael dos Santos Saraiva 
rafaels...@gmail.com wrote:

 Hi

 It's possible in Asterisk 1.8 enable verbose only in one context or
 extension?

 thanks

 Att,
 *Rafael dos Santos Saraiva*
  http://br.linkedin.com/pub/rafael-saraiva/52/aab/230

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Re: [asterisk-users] VoiceMail Issue

[Carlos Rojas]

Hi

Could you send us the logs from the asterisk?

Carlos


On Sat, Mar 8, 2014 at 4:03 AM, Phil Daws ux...@splatnix.net wrote:

 Any ideas on why this may not be working please ?

 - Original Message -
 From: Phil Daws ux...@splatnix.net
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Friday, 28 February, 2014 5:39:54 PM
 Subject: [asterisk-users] VoiceMail Issue

 Hello,

 am attempting again to resolve an issue with multi-tenancy and the
 forwarding to VMs between mailboxes.  If in a multi-tenancy environment one
 uses custom contexts ie.

 [a1-ext1](a1)
 mailbox=101@a1

 and the associated voicemail.conf entry:

 [a1]
 101 = 1234,My User 1,ad...@email.com,,tz=eastern|imapuser=ad...@email.com
 |imapfolder=Inbox
 102 = 1234,My User 2,ad...@email.com,,tz=eastern|imapuser=ad...@email.com
 |imapfolder=Inbox

 now if a message is left in mailbox 101 and the user attempts to forward
 the message to mailbox 102 Asterisk responds that mailbox 102 is not found
 in context default!  One can add:

 searchcontexts=yes

 but that means each mailbox must have a unique number which goes against
 being able to use custom contexts.  I thought by specifying the following
 would fix that:

 exten = 7999,1,VoiceMailMain(${CALLERID(num)}@a1) ; Direct mail retrieval
 exten = 7999,n,Hangup()

 but it does not.  Have tried many ways to resolve but cannot find a
 resolution.

 Any ideas please as would like to get this working ?

 Thank you.



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Re: [asterisk-users] Integration with outlook

[Carlos Rojas]

Hi

Yes, there is, I am using

http://outcall.sourceforge.net/

it's opensource.





On Tue, Jan 28, 2014 at 2:13 PM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hello;

 Is there a method way to be able to dial the phone number through
 asterisk from the outlook email contact?

 Regards
 Bilal

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Re: [asterisk-users] IAX and Variables

[Carlos Rojas]

I thunk so


Let me see
-Original Message-
From: Mikhail Lischuk mlisc...@itx.com.ua
Sender: asterisk-users-bounces@lists.digium.comDate: Tue, 08 Oct 2013 01:08:22 
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAX and Variables

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Re: [asterisk-users] Checking messages from outside the network

[Carlos Rojas]

Are talking about of prepend message?

Because for listening the messages, you can use VoiceMailMain

Carlos Rojas


On Wed, Sep 11, 2013 at 11:37 AM, jg webaccou...@jgoettgens.de wrote:

 Have you considered using VoiceMailMain()?

 jg


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Re: [asterisk-users] VM notification to multiple email recipients

[Carlos Rojas]

Hi

You can do this,
http://mike.eire.ca/2012/02/03/asterisk-1-8-vm-multiple-emails/

If you are using asterisk 1.8


On Wed, Sep 11, 2013 at 1:55 PM, Mike Diehl mdiehlena...@gmail.com wrote:

 Hi all,

 I've got a user who wants to receive voicemail notifications at two
 different email addresses.  I could probably setup an alias in
 /etc/aliases, but then I'd have to manage that across multiple servers,
 which I don't want to do.

 Is there a way I can tell Asterisk to send to multiple addresses?

 Mike

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

[Carlos Rojas]

Hi

You should install something like fail2ban

Regards


On Sun, Aug 18, 2013 at 5:41 PM, Ira i...@extrasensory.com wrote:

  Hello Asterisk-users,

 [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
 [2013-08-18 05:56:34] NOTICE[17089][C-00a9] chan_sip.c:
Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx
 ;tag=7b909220

 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.

 -- Ira

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Re: [asterisk-users] Freeswitch with Digium T316 timed out, T316 timed out

[Carlos Rojas]

My friend,

You are in a wrong list, this an asterisk list, you should to be in
freeswitch list

Kind Regards


On Thu, Aug 8, 2013 at 10:39 AM, Rajat toshniwal 
rajat.toshni...@tekmindz.com wrote:

 **

 Hi

 I am trying to deploy freeswitch with Digium TE121 card for my office
 setup, but it is continuously showing Signaling is up and channels are
 down except D channel.
 Our Architecture is like
 We have freeswitch installed with libpri1.4 and Dahdi.
 I am from India and here we are having E1 trunk.

 Dahdi Configuration is

 cat system.conf
 # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug  7 19:39:07 2013
 # If you edit this file and execute /usr/sbin/dahdi_genconf again,
 # your manual changes will be LOST.
 # Dahdi Configuration File
 #
 # This file is parsed by the Dahdi Configurator, dahdi_cfg
 #
 # Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER)
 span=1,1,0,ccs,hdb3,crc4
 # termtype: te
 bchan=1-15,17-31
 dchan=16
 echocanceller=mg2,1-15,17-31

 # Global data

 loadzone= uk
 defaultzone= uk



 cat modules
 # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules)
 on Wed Aug  7 19:37:48 2013
 # If you edit this file and execute /usr/sbin/dahdi_genconf again,
 # your manual changes will be LOST.
 wcte12xp
 # Xorcom Astribank Devices
 xpp_usb


 dahdi_hardware
 pci::02:08.0 wcte12xp+d161:8000 Wildcard TE121

 dahdi_scan
 [1]
 active=yes
 alarms=OK
 description=Wildcard TE121 Card 0
 name=WCT1/0
 manufacturer=Digium
 devicetype=Wildcard TE121 (VPMOCT032)
 location=PCI Bus 02 Slot 09
 basechan=1
 totchans=31
 irq=0
 type=digital-E1
 syncsrc=1
 lbo=0 db (CSU)/0-133 feet (DSX-1)
 coding_opts=AMI,HDB3
 framing_opts=CCS,CRC4
 coding=HDB3
 framing=CCS/CRC4


 Card is properly installed and recognized by Dahdi

 Freetdm is compiled with libpri and configuration is like
 cat /usr/local/freeswitch/conf/freetdm.conf
 [general]
 cpu_monitor = yes
 cpu_monitoring_interval = 2000 ; monitor usage every 2 seconds
 cpu_set_alarm_threshold = 90 ; whenever 90% of global CPU usage is
 reached, trigger the alarm.
 cpu_reset_alarm_threshold = 80 ; when the CPU usage decreases at 80%,
 clear the alarm.
 cpu_alarm_action = reject,warn ; Start rejecting calls when the CPU
 alarm is triggered and also print warnings.

 [span zt myDAHDISpan]
 trunk_type = E1
 group = g1
 b-channel = 1-15
 d-channel = 16
 b-channel = 17-31


 cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
 configuration name=freetdm.conf description=FreeTDM Configuration
 settings
 param name=debug value=1/
 /settings
 libpri_spans
 span name=myDAHDISpan
 !-- Log Levels: none, alert, crit, err, warning, notice, info, debug --
 param name=switch value=euroisdn/
 param name=node value=cpe/
 param name=dialect value=q931/
 param name=debug value=all/
 param name=dialplan value=XML/
 param name=context value=public/
 param name=l1 value=alaw/
 /span
 /libpri_spans
 /configuration


 Freeswitch logs are showing

 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c10][1:10]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c11][1:11]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c12][1:12]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c13][1:13]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c17][1:17]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c18][1:18]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c19][1:19]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c20][1:20]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c21][1:21]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c22][1:22]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c23][1:23]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c24][1:24]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c25][1:25]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c26][1:26]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c27][1:27]
 -- T316 timed out, resending RESTART request
 2013-08-08 15:38:18.633847 

Re: [asterisk-users] Mysql Support int Asterik-11

[Carlos Rojas]

Hi

Asterisk 1.6 and old versions, were using asterisk-addons, since asterisk
1.8 asterisk addon, is included in the asterisk code, you must select it in
menu select.

Kind Regards

Carlos


On Wed, Jul 24, 2013 at 8:36 AM, Prashant Abhang 
abhang_prash...@yahoo.co.in wrote:

 I have done using odbc..

 but I was curious to know ..whether it directly possible using mysql so I
 can avoid installation of unixodbc pkg.



 
 Thanks  Regards,
 Prashant Abhang

   --
  *From:* Thorsten Göllner t...@ovm-group.com
 *To:* Prashant Abhang abhang_prash...@yahoo.co.in; Asterisk Users
 Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com

 *Sent:* Wednesday, 24 July 2013 5:57 PM
 *Subject:* Re: [asterisk-users] Mysql Support int Asterik-11

  Why not use ODBC?

 Am 24.07.2013 13:41, schrieb Prashant Abhang:


  Hi,

 I was having question about mysql driver support ( not odbc).

 Do we still need the asterisk-add-on to be installed for mysql support.
 If yes, Which version should be used and from where I should get it?

  Thanks in adavance.
  
 Thanks  Regards,
 Prashant Abhang





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Re: [asterisk-users] Asterisk 1.8 Service: -r does not give CLI

[Carlos Rojas]

Not it didn't,

Did you execute asterisk -r
or /usr/sbin/asterisk -r ?

If not working did you execute

asterisk -gc  ?


Kind Regards


On Mon, Jul 22, 2013 at 10:41 AM, Meadows Hoa meadows_...@yahoo.com wrote:

 We have Asterisk1.8.11 and can not move to a newer version right now. But
 when we run Asterisk as a service, the -r option does not result in giving
 the CLI prompt? Did the option to get the CLI change?

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Re: [asterisk-users] Asterisk offline compiling with get_mp3_source.sh

[Carlos Rojas]

Hi

You must copy the directory mp3, to the addons directory, where you put the
source asterisk code, and recompile it, again.

Kind Regards




On Mon, Jul 15, 2013 at 9:25 AM, leonardo collantes leonardo07...@gmail.com
 wrote:

 I need to make a Asterisk 18.0's offline compiling,  SVN mp3 support
 sources downloading does't particulary works cause my asterisk is in an
 isolated network with NO network access whatsoever, I ve read this thread (
 http://lists.digium.com/pipermail/asterisk-users/2013-June/279298.html)
 but I 'm not understading one thing, because I download the file and run
 the script but there is no asterisk-contrib-mp3.tar.gz in my tmp folder


 --- contrib/scripts/get_mp3_source.sh.orig2013-06-04 12:41:08.222602824 
 +0200
 +++ contrib/scripts/get_mp3_source.sh 2013-06-04 12:40:45.218602846 +0200
 @@ -9,6 +9,15 @@
  exit 1
  fi

 +LOCAL_COPY=/tmp/asterisk-contrib-mp3.tar.gz
 +if [ -f ${LOCAL_COPY} ]; then
 +echo ***
 +echo Found ${LOCAL_COPY} - unpacking it, not downloading
 +echo ***
 +tar xzf ${LOCAL_COPY}
 +exit 0
 +fi
 +
  svn export http://svn.digium.com/svn/thirdparty/mp3/trunk addons/mp3 $@

  exit 0




 and i don't know what to do with the mpglib file




 asterisk (1:1.8.13.1~dfsg-3) mpglib Summary

  addons/mp3/MPGLIB_README |   39
  addons/mp3/MPGLIB_TODO   |2
  addons/mp3/Makefile  |   24
  addons/mp3/README|1
  addons/mp3/common.c  |  267 ++
  addons/mp3/dct64_i386.c  |  335 +++
  addons/mp3/decode_i386.c |  153 +++
  addons/mp3/decode_ntom.c |  219 +
  addons/mp3/huffman.h |  332 +++
  addons/mp3/interface.c   |  323 +++
  addons/mp3/layer3.c  | 2029 
 +++
  addons/mp3/mpg123.h  |  132 +++
  addons/mp3/mpglib.h  |   75 +
  addons/mp3/tabinit.c |   81 +
  14 files changed, 4012 insertions(+)




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Re: [asterisk-users] asterisk -rx core show channels + time

[Carlos Rojas]

Hi

You can do,

core show channels verbose


Kind Regards


On Thu, Jun 20, 2013 at 6:45 PM, Joseph syscon...@gmail.com wrote:

 When I type: asterisk -rx core show channels
 I usually get
 Channel  Location State   Application(Data)
   SIP/pstn--03 7807574622@internal: Up
  Dial(SIP/77807574622@pstn-9998
 SIP/pstn-9998-03 (None)   Up  AppDial((Outgoing Line))

 Is there a way to pull information about time the channel started?

 --
 Joseph

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Re: [asterisk-users] dCAP study recommendations

[Carlos Rojas]

Hi,

If you read, O'Reilly - Asterisk - The Definitive Guide - 3rd Edition, you
should be ready for take the test.

Of course, you must read voip-info too.

Carlos Rojas
Dcap 2171


On Fri, Jun 7, 2013 at 2:20 PM, Michael Gilleran mgille...@realtyim.comwrote:

  Greetings. Anyone have any recommendations for studying for the dCAP
 Certification? Other than the expensive Digium courses, there doesn’t seem
 to be anything online.

 ** **

 Thanks,

 ** **

 ** **

 *Michael Gilleran  *

 

 ** **

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Re: [asterisk-users] monitoring Asterisk 1.8

[Carlos Rojas]

I'm using opennms and It's working fine.





On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote:

 Hello,

 i'm looking for suggestions to monitor Asterisk Server? I installed Nagios
 but no success, I do prefer not to install any web server on the server
 running Asterisk.


 Thanks in advance.
 -Motty

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Re: [asterisk-users] monitoring Asterisk 1.8

[Carlos Rojas]

http://opennms.org/wiki/Installation:Yum


On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote:

 I'm using opennms and It's working fine.





 On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote:

 Hello,

 i'm looking for suggestions to monitor Asterisk Server? I installed
 Nagios but no success, I do prefer not to install any web server on the
 server running Asterisk.


 Thanks in advance.
 -Motty

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Re: [asterisk-users] monitoring Asterisk 1.8

[Carlos Rojas]

You can use queue-stats
http://www.asternic.org/stats/demo/

they has a free version




On Thu, May 9, 2013 at 4:12 PM, motty cruz motty.c...@gmail.com wrote:

 Thanks for your help; I just want to monitor the queue, calls on hold
 average time, incoming out going call, I only want to monitor Asterisk, not
 the server Asterisk in running on.

 thanks,
 -Motty


 On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote:

 http://opennms.org/wiki/Installation:Yum


 On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote:

 I'm using opennms and It's working fine.





 On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote:

 Hello,

 i'm looking for suggestions to monitor Asterisk Server? I installed
 Nagios but no success, I do prefer not to install any web server on the
 server running Asterisk.


 Thanks in advance.
 -Motty

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Re: [asterisk-users] problem

[Carlos Rojas]

Hi

Are you sure  that your hard drive sda, is ok?

Looks like your hard drive is broken.

On Wed, Feb 6, 2013 at 10:30 AM, brahim abidar abidarbah...@gmail.comwrote:

 Hi every body;

  I want to intall  some softwars working with my Asterisk server and I get
 these erreurs :

 *
 error: cannot seek `/dev/sda'.
 error: cannot seek `/dev/sda'.
 error: cannot seek `/dev/sda'.
 /usr/sbin/grub-probe: error: cannot seek `/dev/sda'.
 dpkg: error processing grub-pc (--configure):
  subprocess installed post-installation script returned error exit status 1
 *

 Please can any one let me know how can I resolve this problem;
 --
 *
 **Élève Ingénieur INE2 à l'Institut National des Postes et
 Télécommunications * *INPT - Rabat - Maro*c *

 *
 * * *Responsable de la cellule Asterisk au **Club Electronique et
 Systemes Embarqués de l'INPT*
 *Membre du projet  ilearn, SIFE INPT* *
*
 * Tel : +212642398782
*

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Re: [asterisk-users] g723 transcoding

[Carlos Rojas]

Hi

Look at it this link

http://asterisk.hosting.lv/


Kind Regards

On Thu, Jan 24, 2013 at 10:34 AM, Richard Kenner ken...@gnat.com wrote:

 It appears that there are no transcoders from g723 to anything else in
 Asterisk 10.7.1.  Does anybody know how to fix that?

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Re: [asterisk-users] - configure ring group

[Carlos Rojas]

Maybe,

You can do that, with queues, and ringall strategy.

On Wed, Dec 5, 2012 at 4:53 PM, Leandro Dardini ldard...@gmail.com wrote:

 You can dial all the extensions at once, putting all them in the dial
 string, separated by . There is no other method.

 Leandro

 2012/12/5 Paolo De Michele pa...@paolodemichele.it

  hi all,

 I want have an information about ring group in asterisk (1.8.16 - centos
 6.3)
 I have configured skypeforasterisk for incoming call to one extension and
 it works

 now,my chan_skype.conf is:

 [general]

 default_user=user-skype

 [user-skype]
 secret=x
 context=from-skype
 exten=
 disallow=all
 allow=ulaw
 allow=alaw

 my extensions.conf:

 [from-skype]

 exten = ,1,Verbose(2,Incoming Skype Call)
same = n,Answer()
same = n,Dial(SIP/1000SIP/2000SIP/3000,30)
same = n,Playback(useris-curntly-unavail)
same = n,Hangup()

 at right time the internal ring are 1000, 2000 and 3000
 I have the extension from 1000 to 1005, 2000 to 2005 and from 3000 to 3005
 I can ring him all? I can group the configuration into a single string?

 let me know something
 thanks in advance




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Re: [asterisk-users] Allowing peers from specific subnet only

[Carlos Rojas]

Hello

In SIP.find you can to use

Deny=0.0.0.0/0.0.0.0
Permit=192.168.1.25/255.255.255

Regards
On Nov 19, 2012 7:12 AM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hi;

 How I can make my configuration to allow the sip phones only from specific
 IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be
 allowed to connect for asterisk?

 In other words, in addition to be authenticated based on the username and
 password, it is required that the IP address of the Phone to be from this
 range. How?

 Regards
 Bilal

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Re: [asterisk-users] multitenanat third party app

[Carlos Rojas]

Hi

You will need change the names for your extensions

101-company_a
102-company_a

ETC



On Wed, Oct 31, 2012 at 2:23 PM, Darin Iv adari...@gmail.com wrote:
 Is it possible to bul multitenant system using some third party opensouce
 application My design is like this.

 Company A:
 Context Company_A
 IVR Company A
 Extensions: 101,102,103,104 etc.

 Company B:
 Context Company_B
 IVR Company B
 Extensions: 101,102,103,104 etc.

 Company C:
 Context Company_C
 IVR Company
 Extensions: 101,102,103,104 etc.


 Company D:
 Context Company_D
 IVR Company D
 Extensions: 101,102,103,104 etc.


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Re: [asterisk-users] blocking incoming call - asterisk 1.8

[Carlos Rojas]

Hello

Yes, has a berckeley database, wirh function blackllist

Regards
On Oct 9, 2012 12:51 AM, Joseph syscon...@gmail.com wrote:

 Can someone refresh my memory how blocking incoming call works based on
 caller ID in Asterisk 1.8?
 If I remember correctly in asterisk 1.4 it was possible to block caller ID
 from the command line, asterisk had some internal database I think.

 --
 Joseph

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Re: [asterisk-users] Asterisk 1.8.10

[Carlos Rojas]

Hello

You should be modify the volume  in the file, there are several
software  for that, like  wavepad .

Regards

On Mon, Oct 1, 2012 at 2:52 PM, Danny Nicholas da...@debsinc.com wrote:
 AFAIK,  there is still not a MOH volume control.  What I did was to take my
 moh wav files and run them through sox like this
  $ cp -iv macroform-cold_day.wav macroform-cold_day_orig.wav
 $ sox -v 0.9 macroform-cold_day_orig.wav macroform-cold_day.wav

 This produces a file 90 percent as loud.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
 Sent: Monday, October 01, 2012 1:44 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] Asterisk 1.8.10

 I can't find a clear procedure to lower musicOnHold volume!

 Any suggestions?

 Hereis my music.conf file


 [default]
 mode=files
 directory=moh

 Thanks in advance!


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Re: [asterisk-users] Remote SIP Extension Best Practices

[Carlos Rojas]

Hi

Ok, I think vpn is good way, but , you can use tls that uses certificates,
and srtp for media encriptatio, in sip protocol.

Regards
On Sep 29, 2012 12:59 PM, Chris Nighswonger cnighswon...@foundations.edu
wrote:

 On Sat, Sep 29, 2012 at 12:41 PM, Carlos Rojas crt.ro...@gmail.com
 wrote:
  Hello.
 
  Vpn is good idea, is more secure, you can use tls with srtp as well.
 
  Are you using asterisk 1.8? Right?

 Asterisk 10.7.0

 Kind Regards,
 Chris

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Re: [asterisk-users] Hangup not detected

[Carlos Rojas]

Hello

In indications.com are the tones for several countries
On Sep 18, 2012 4:34 AM, Mehdi Rahimi mrm.ci...@gmail.com wrote:

 Hi AJS,

 Thank you for your reply , I am using this in IRAN so please guide me
 what to do and and explain me more.
 Look forward to hearing from your side.
 Regards,
 Mehdi

 On Tue, Sep 18, 2012 at 11:28 AM, A J Stiles
 asterisk_l...@earthshod.co.uk wrote:
  On Tuesday 18 September 2012, Satria Anamarta wrote:
  Hi,
  I just realize in these few days there are many calls that already
 hangup
  but not detected by Asterisk.
  Those calls occupy PSTN lines and need to be manually terminated through
  Flash Operation Panel or phycally disconnect the PSTN lines.
  This never happen before but as long as I can remember, there are no
 change
  in configuration.
 
  Any ideas how to solve this?
 
  If you are using analogue phone lines in some country that uses a
 British-
  style telephone system  (line wires called A and B, not tip and
 ring;
  polarity reversal before ringing; double ring on incoming call),  then by
  design only the calling party can terminate a call once established.  If
  someone rings you and you hang up but they stay on the line, you will
 still be
  connected to them if you later pick up the phone -- the call is only
  disconnected once the calling party hangs up.
 
  Asterisk is aware of this, and takes steps to mitigate it.  The fix is
 simply
  to make sure you specify the correct country in your DAHDI configuration.
 
  --
  AJS
 
  Answers come *after* questions.
 
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Re: [asterisk-users] Check for the voicemail

[Carlos Rojas]

Hello

Check voicemail.conf

maxmsg = 100

And change it.




On Tue, Aug 21, 2012 at 12:52 PM, Danilo Dionisi
dionisi.dan...@gmail.com wrote:
 I'm sorry, I haven't been clear.
 I do not have to check the inbox on Asterisk, but I have to check the free
 space on a particular mailbox of Exchange software.
 It's possible with the pair Asterisk-Sendmail?

 Il 21/08/12 18:45, Danny Nicholas ha scritto:

 Assuming that you are using the standard 100 message limit, just check for
 INBOX/MSG0100.txt and send the message.


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Re: [asterisk-users] Hosted Softswitch Integration

[Carlos Rojas]

Hello

I think you must change

type = peer
insecure=invite,port
qualify=yes ; for monitor the ip



Regards

On Fri, Aug 17, 2012 at 2:11 PM, Selecstine Bucci Anukwu
buchal...@gmail.com wrote:
 Hello Everyone,

 We are trying to integrate a hosted soft-switch to an Asterisks server and
 the error received on the Softswitch end is decline 603

 The change that we made is to add the Softswitch IP in the SIP configuration
 file, see below


 [from-trunk]
 host=66.77.199.205
 type=user
 nat=yes
 insecure=very
 dtmfmode=rfc2833
 context=from-trunk
 canreinvite=no
 disallow=all
 allow=ulaw
 allow=gsm
 allow=g729


 On the attempt to integrate to the asterisks server nothing is seen in the
 asterisk log
 There is something we might not be doing well. can somebody please help.

 Regards

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Re: [asterisk-users] Segmenting A Configration File

[Carlos Rojas]

Hi

Have you seen thirdlane?
Thirdlane has a multitenant version.

Regards
On Aug 11, 2012 11:11 AM, Carlos Alvarez car...@televolve.com wrote:

 On Sat, Aug 11, 2012 at 3:16 AM, Kannan vasdevelo...@gmail.com wrote:

 I am planning a multi-tenant VoIP services system with Asterisk, using
 configuration tweaks. Having all the tenant configurations in one
 configuration file is overwhelming. I would like to segment the
 configuration files and include them in the main configuration file. Is it
 possible?

 For e.g. I would like to have the main extenstions.conf file to include
 tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy
 to manage the configurations of each tenant.


 We put each tenant's sip and extensions config files in
 /etc/asterisk/accounts and then do an include for that directory in the
 main files.

 We keep all the voicemail.conf in one because changes to passwords will
 NOT be saved to included files.  We used to use includes for voicemail but
 that meant no password changes.

 The main file has a list of all phone numbers in the system in numerical
 order where we set the account name, and then we send them to the proper
 context like this:

 exten = 12015551212,1,Set(CDR(accountcode)=johnsmith)
 

 exten = _X.,n(cont),Goto(${CDR(accountcode)}#did,${EXTEN},1)

 There's a bunch of other stuff in there where we do line counting and such.

 --
 Carlos Alvarez
 TelEvolve
 602-889-3003



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Re: [asterisk-users] Voice Mail beep / tone detection

[Carlos Rojas]

Hello

You will need to do, something like

[outbound]

exten = s,1,NoCDR

exten = s,n,AMD

exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)

exten = s,n(mach),WaitForSilence(2500)

exten = s,n,Playback(message-when-machine)

exten = s,n,Hangup

exten = s,n(humn),WaitForSilence(500)

exten = s,n,Playback(message-when-human)

exten = s,n,Hangup


On Sun, Aug 5, 2012 at 12:52 PM, tahir almas ta...@ictinnovations.com wrote:
 Though asterisk support AMD which is based on silence detection but I did
 not found support of  tone / beep detection in asterisk to record a voice
 message for answering machines after detecting tone

 Will appreciate any help in this regard

 Best Regards
 Tahir Almas

 Managing Partner
 ICT Innovations
 http://www.ictinnovations.com
 Leveraging open source in ICT

 Unified Communication Telemarketing Software
 http://www.ictbroadcast.com


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Re: [asterisk-users] any working calling card solution open source

[Carlos Rojas]

Hello

a2billing works fine

Regards

On Mon, Jul 16, 2012 at 1:47 PM, Goke M Aruna gok...@gmail.com wrote:
 hi all,

 Can someone give me information on any open source asterisk calling card
 solution?
 I have laid my hands on astpp, astcc, asterisk-prepaid-0.3.1, agi-ccard.agi
 without luck.
 I guess my problem is Asterisk-perl

 I will be glad for a quick response.

 Regards

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Re: [asterisk-users] weird dect beheaviour multiple handsets

[Carlos Rojas]

Hello

Is your server behind nat? This problems sounds me nat problems.


Regards

On Thu, Jul 12, 2012 at 7:53 AM, Roland o/d Akker aster...@rolandow.com wrote:
 I have this very specific problem with two dect sets. Problem that I have is
 one-way audio, in this very rare situation.

 I am calling with a Siemens N510 with C610 handset to Panasonic KX-TGP500
 with KX-TPA50 handset. This gives me problems when I am calling to a SIP
 account that is configured to ring all handsets. Then when one handset
 answers, I only hear the panasonic, but they don't hear me.

 When I call to an extension that is configured to ring only one handset, I
 don't have this problem.

 When I use panasonic on both sides, or Tiptel - Panasonic, I don't have
 this problem as well.

 I am breaking my head what this could be. Any idea's?

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Re: [asterisk-users] New to Asterisk

[Carlos Rojas]

Hello

http://www.voip-info.org/wiki/view/Asterisk

I prefer asterisk under linux sistem works better.


Regards

On Sun, Jun 17, 2012 at 12:28 PM, Jim Schultz jimschultz...@gmail.comwrote:

 Greetings,

 I am interested in learning more ablout Asterisk. Is there a recommended
 link for getting started. Can I set up an Asterisk server on my Win 7
 local host ?? Is this what I need to do or is there another way of becoming
 familiar with the Asterisk product ?

 Any help and guidance for a new user is much appreciated ?

 Jim

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Re: [asterisk-users] Line noise/hiss on Openvox A400P card on FXO

[Carlos Rojas]

Hello

Are you using a amd server?

Sometimes openvox doesn't work fine with amd processor

Regards
On Mar 1, 2012 2:07 PM, Dave Platt dpl...@radagast.org wrote:

  5. Placing ferrite cores on the phone cables.

 Do either of the phone lines in question have DSL on them?

 If so, a ferrite core (which will block common-mode RF
 signals) probably won't help much, if at all.  DSL is a
 differential-mode signal, and its frequency content starts
 down in the tens of kHz.  Ferrite cores are usually intended
 to block much higher frequency interference, and won't have
 enough inductance to help much with DSL signals.

 What I would suggest, is that you get yourself a couple
 of DSL microfilters... plug them into the A400P FXO
 ports, and plug the lines into the filters.  These sorts
 of filters are designed specifically to block DSL differential-
 mode signals from getting into analog-phone circuits, and
 they will also be fairly effective against other forms
 of low-frequency-RF noise.



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[asterisk-users] Virtual Server

[Carlos Rojas]

Hello everybody

someone in this list, has installed asterisk, in a virtual server like
 proxmox? I'm thinking  install some asterisk servers in a machine dell
xeon 64 processor, but I'm not sure, about virtual Server software.

I heard, about proxmox, but I don't know if works fine.

Regards

Carlos
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Re: [asterisk-users] asterisk problem sip

[Carlos Rojas]

Hi,

Thanks a lot, for your help, I found, much iformation about that will test
the bugs, for asterisk

Regards

carlos
On Sat, Jan 14, 2012 at 3:35 PM, Alec Davis siva...@paradise.net.nz wrote:

 **
 Sounds like you've run in to a deadlock problem.

 Running 'core show locks' at the asterisk CLI, will show you that you
 have a lock, but debugging this is fun.
 To be able to use 'core show locks' you need 'DEBUG THREADS' and 'DONT
 OPTIMIZE' enabled in 'make menuslect'

 However, 1.6.22 is the lastest.

 Alec Davis


  --
 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Carlos Rojas
 *Sent:* Saturday, 14 January 2012 3:37 p.m.
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] asterisk problem sip

 Hi everybody

 I have been presenting a periodic problem, do not know if anyone listed
 has happened something similar,I'm using the asterisk, asterisk-1.6.2.13,
 in different locations works well, but every so often fails, hangs on
 Asterisk server or simply asterisk, SIP requirements do not answer,
 apparently unaware of the dialplan as well as voicemails, etc, but keeps
 the registry, apparently the service is up but theusers, call and doesn't
 make calls.


 Asterisk is restarted, and solve the problem, I think it may be a bug in
 asterisk, someone has had a similar problem?


 Regards

 Carlos


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[asterisk-users] asterisk problem sip

[Carlos Rojas]

Hi everybody

I have been presenting a periodic problem, do not know if anyone listed has
happened something similar,I'm using the asterisk, asterisk-1.6.2.13, in
different locations works well, but every so often fails, hangs on Asterisk
server or simply asterisk, SIP requirements do not answer, apparently
unaware of the dialplan as well as voicemails, etc, but keeps the
registry, apparently the service is up but theusers, call and doesn't make
calls.


Asterisk is restarted, and solve the problem, I think it may be a bug in
asterisk, someone has had a similar problem?


Regards

Carlos
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Re: [asterisk-users] dialplan - dial command - custom ringtone

[Carlos Rojas]

Hello

Do you use hard phone or softphone?

In many ip phones you can change the ring tones or use w option in Dial
command

Regards
On Jan 3, 2012 4:08 AM, Qqblog Qqblog qqb...@ymail.com wrote:

 i could add r option in dial command. this will generate a ringtone
 during connection. could i change this default ringtone?

 i tried indications.conf but not success.

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Re: [asterisk-users] IAX2 woes

[Carlos Rojas]

Hello

Asterisk only says that the iax2 channel don't work maybe you look the
iax.conf. you trunk. Is iax I think

Regards
On Dec 29, 2011 6:49 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:

 Hello all,

 I attempted to make a couple of outbound calls this morning and always got
 the busy tone.  I checked the Asterisk console and was greeted with:

 [Dec 29 11:29:22] WARNING[12039]: app_dial.c:2218 dial_exec_full: Unable
 to create channel of type 'IAX2' (cause 20 - Unknown)
   == Everyone is busy/congested at this time (1:0/0/1)

 I proceeded to restart Asterisk and dialed the same number again and it
 worked without fault. What could cause this type of error and is there any
 way to auto-remediate when it does arise ?

 voip*CLI core show version
 Asterisk 10.0.0 built by root @ voip.my.server on a x86_64 running Linux
 on 2011-12-19 16:16:46 UTC
 --
 Thanks, Phil


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Re: [asterisk-users] Monitor Command Records separate channales

[Carlos Rojas]

Hello,

Do you use monitor?, because in asterisk 1.4 to new versions, It's use
mixmonitor, in asterisk 1.2 had this mistake.

Regards

On Wed, Dec 28, 2011 at 10:11 PM, Steve Edwards
asterisk@sedwards.comwrote:

 Un-top-posting, snarky comments inline...


 On Wed, 28 Dec 2011, Faraj Khasib wrote:

  I am trying to record Call, but when the call is done I have one file but
 the conversation inside it is separate into calls conversation and receiver
  its single file but separate recording, How can I make it mixed
 together so the conversation will be normal?


 (I don't understand. How can you have separate recordings in a single
 file?)


 On Wed, 28 Dec 2011, Faraj Khasib wrote:

  I installed SOX( it was not installed before). Will that solve my
 problem? if not what are the parameter for the mixMonitor Command this is
 how I use Monitor


  exten=6500,2,Monitor(...)


  is Mix Monitor will have the same?


 (You want me to guess if installing sox will solve your problem?)

 (Too lazy to look up the mixmonitor() command?)


 On Wed, 28 Dec 2011, Faraj Khasib wrote:

  Asterisk 1.6.2 but sox I don't know but now it is the latest version, my
 problem is not mixing  It's the same file but inside that file two
 seperate records first callers then reciever


 (Finally we get some details...)


 On Wed, 28 Dec 2011, Faraj Khasib wrote:

  Can u plz tell me how , I forgot how to run asterisk cli


 (Lazy or in over his head?)


 On Wed, 28 Dec 2011, Faraj Khasib wrote:

  My call happens with a queue , there is no full file but there is queue
 and queue is useless, can u give me unix command to search all log files
 and print moniter line?


 (Don't understand the question and seeing the lazy/unqualified thing
 again.)


 On Wed, 28 Dec 2011, Faraj Khasib wrote:

  I already searched using grep for the monitor word ... It doesn't exists


 (Don't have a lot of confidence in this statement.)


 On Wed, 28 Dec 2011, Faraj Khasib wrote:

  but i tiried these commands and I didnt find anything about Monitor


  [root@c-24-1-71-68 asterisk]# grep -R 'Monitor' *


  [root@c-24-1-71-68 asterisk]# grep -R 'monitor' *


 (Should brush up on grep's command line parameters. I wonder what '*'
 evaluates to. I hope he isn't really logged in as root)

 On Wed, 28 Dec 2011, Faraj Khasib wrote:

  It got stuck ...


 (I wonder what this means. What is he talking about?)


 On Wed, 28 Dec 2011, Faraj Khasib wrote:

  I attached log, but there is nothing unusual in it ...all normal ...


 (No file attached. Maybe he should read the error message the list manager
 returned. Little confidence in his assessment as 'normal')

 Please take a moment to learn list etiquette.

 1) Please don't top post.

 2) Please don't ask questions you could easily google yourself.

 3) Please learn basic Unix commands like 'grep'.

 4) Please take the time to form unambiguous questions.

 5) Please include sufficient detail so we don't have to keep guessing what
 is going on.

 It appears (from your 3rd post) that your problem is that the monitor()
 application is concatenating both 'legs' of the call into a single file --
 meaning that when you play the single recorded file you hear the entire
 conversation from the caller's side and then you hear the entire
 conversation from the callee's side. Kind of like:

 Callee) Hello?

 Callee) Fine, but I really have no clue what I'm doing.

 Callee) Never heard of it. Besides all these schmucks on the AU list like
 reading basic questions and spoon-feeding me the answers.

 Callee) There's a quota?

 Callee) How many questions do I have left?

 Callee) Steve?

 Callee) Hello?

 Callee) Hmmm. I must have a problem with my upstream provider...

 Caller) Hey Faraj, how ya doing?

 Caller) Sorry to hear that. Have you ever tried Google?

 Caller) Hmmm. Have you burned through your newbie question quota yet?

 Caller) Yep. It's not set in stone, but if you keep at it without showing
 you're putting in any effort, everybody will figure you out and ignore you.

 Instead of:

 Callee) Hello?

 Caller) Hey Faraj, how ya doing?

 Callee) Fine, but I really have no clue what I'm doing.

 Caller) Sorry to hear that. Have you ever tried Google?

 Callee) Never heard of it. Besides all these schmucks on the AU list like
 reading basic questions and spoon-feeding me the answers.

 Caller) Hmmm. Have you burned through your newbie question quota yet?

 Callee) There's a quota?

 Caller) Yep. It's not set in stone, but if you keep at it without showing
 you're putting in any effort, everybody will figure you out and ignore you.

 Callee) How many questions do I have left?

 Callee) Steve?

 Callee) Hello?

 Callee) Hmmm. I must have a problem with my upstream provider...

 This would be a novel problem since in over 8 years of reading this list
 I've never seen anybody else report it.

 Does this happen with all calls or only calls that are queued to an agent?

 Are you fiddling with MONITOR_EXEC or 

Re: [asterisk-users] Interesting attack tonight fail2ban them

[Carlos Rojas]

Hello,

Do you set up, your logrotate in /etc/asterisk ?
Do you test that your fail2ban work fine?

Regards

On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis mdup...@ocg.ca wrote:

  I happened to be in the cli tonight as some (208.122.57.58) initiated a
 simple attack - just trying to make long distance calls from outside
 context.  Although harmless, this went on for several minutes as the idiot
 just used up my bandwidth with SIP messages.  Here's and example:

 [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035
 handle_request_invite: Call from '' to extension '6442032987219' rejected
 because extension not found.
 [2011-12-28 22:53:44] NOTICE[9635]: chan_sip.c:14035
 handle_request_invite: Call from '' to extension '7442032987216' rejected
 because extension not found.
 [2011-12-28 22:53:46] NOTICE[9635]: chan_sip.c:14035
 handle_request_invite: Call from '' to extension '8442032987216' rejected
 because extension not found.
 [2011-12-28 22:53:48] NOTICE[9635]: chan_sip.c:14035
 handle_request_invite: Call from '' to extension '008442032987215' rejected
 because extension not found.
 [2011-12-28 22:53:50] NOTICE[9635]: chan_sip.c:14035
 handle_request_invite: Call from '' to extension '007442032987218' rejected
 because extension not found.
 [2011-12-28 22:53:52] NOTICE[9635]: chan_sip.c:14035
 handle_request_invite: Call from '' to extension '006442032987219' rejected
 because extension not found.
 [2011-12-28 22:53:54] NOTICE[9635]: chan_sip.c:14035
 handle_request_invite: Call from '' to extension '005442032987216' rejected
 because extension not found.
 [2011-12-28 22:53:56] NOTICE[9635]: chan_sip.c:14035
 handle_request_invite: Call from '' to extension '004442032987250' rejected
 because extension not found.

 I thought that it might be worth adding a line to my fail2ban filter, but
 am looking for a hand with the regex.  I have come up with:
 NOTICE.* .*: Call from '' to extension '.*' rejected because
 extension not found

 but I realize that anyone misdialling a valid extension a few times gets
 cut off. Can someone suggest an improvement?  (How could I limit this to 4
 or more digits dialled for example?)

 Thanks!

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Re: [asterisk-users] Client - registers but unreachable

[Carlos Rojas]

Hello,

Your blackberry sip client, works in your wifi network? or by blackberry
internet?
do you set nat=yes if your phone, register by internet?

What is your sip.conf?


Regards

On Wed, Dec 28, 2011 at 11:16 PM, Michelle Dupuis mdup...@ocg.ca wrote:

  I have a softphone I'm trying on a blackberry, that registers on my
 Asterisk, can make outgoing calls, but can't receive calls.

 There is very little traffic with this phone (see debug below - as the
 phone registers), and sip show peers confirms it is unreachable.

 Any suggestions?  Is this just a dumb client or do I need to tweak an
 asterisk setting?

 Thanks

 pbx*CLI sip debug peer 230bb
 Unable to get IP address of peer '230bb'
 The 'sip debug' command is deprecated and will be removed in a future
 release. Please use 'sip set debug' instead.
 pbx*CLI
 -- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800
 [2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 sip_poke_noanswer:
 Peer '230bb' is now UNREACHABLE!  Last qualify: 0

 pbx*CLI sip show peers
 Name/username  HostDyn Nat ACL Port
 Status
 230bb/bob  172.31.254.53D  9653
 UNREACHABLE

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Re: [asterisk-users] how to stop hacking of my server

[Carlos Rojas]

Hello

I use fail2ban, and works fine,


Regards

On Tue, Dec 27, 2011 at 1:54 AM, virendra bhati virbh...@gmail.com wrote:

 Hi list someone is trying to hack my server . Is there any way by whcih I
 can stop hacking of my server except iptables ? I want to stop on the basis
 of sip.conf account only. bcoz I can't apply iptables rules on server it's
 remote server of server provider and we used it for making voip call for
 customers.

 for the time been i have close all sip accounts. but can't stop for more
 then 1 days. I need your help 

 *CLI log:- *
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - Wrong password
 [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
 Registration from '4411 sip:4411@204.152.194.246' failed for
 '62.141.54.169' - 

Re: [asterisk-users] GOIP GSM to SIP Gateway?

[Carlos Rojas]

Hello

It is possible but how do you have the dialplan ?
In your dial plan you can do that

Regards
On Dec 20, 2011 2:40 PM, Matt mhop...@gmail.com wrote:

 Hi,
 Has anyone here any experiencing with linking an Asterisk PBX to a
 GOIP GSM to SIP Gateway?  We've got inbound calls from the GSM network
 working properly, however, outbound calls seem to randomly choose a
 SIM line to use.

 Is there anyway (short of defining dial an 8 from this phone for this
 trunk to this SIM and a 9 from this phone for a trunk to this SIM) to
 get it to use certain SIM cards when calls are made from certain
 phones?

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[asterisk-users] asterisk and heartbeat

[Carlos Rojas]

Hello everybody

I'm setting, heartbeat and asterisk, with rsync, anyone, work them fine?

I've been find any information and saw heatbeat + cysnc2 and heartbeat +
rdbd, any one has worked any these aplications fine?


Best regards
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Re: [asterisk-users] How to monitor SIP Trunk on production server

[Carlos Rojas]

Hello,

Do you saw this solution?

http://linuxnotes.us/


Regards

On Sun, Dec 18, 2011 at 12:26 AM, virendra bhati virbh...@gmail.com wrote:

 Hi List,

 I have asterisk 1.6.2.20 installed at production server, I have 2 SIP voip
 trunk for making outgoing and DID for incoming to server.

 My question is how I can ensure that trunk is not down at production
 server, So how I can monitor it's automatically by making any scripts?

 Any hint will be appreciated

 --

 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


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Re: [asterisk-users] Set Caller Number in E1 PRI ISDN Lines

[Carlos Rojas]

Hello
Did you use callerid(num) in your dial plan?
On Dec 16, 2011 7:38 PM, Kaushal Shriyan kaushalshri...@gmail.com wrote:

 Hi,

 I am having an E1 PRI ISDN Lines with 30 bearer channels and 1 D Channel
 with hundred DIDs (Direct Inward Dialing) numbers attached to a Sangoma PRI
 Card on the server,
 I am using asterisk 1.8.5 on CentOS 5.6.

 How can i configure DIDs so that if i make an outgoing call the DID number
 should go to the caller not the pilot number

 For example

 PRI Numbers Range - 31303000 - 31303099
 Pilot Number - 31303000

 So if i need to set caller number as 31303008 for example and not as
 31303000, is there a way to set this in dial plan (extensions.conf)

 Please guide and let me know if anyone needs more information and have
 questions.

 Regards,

 Kaushal

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[asterisk-users] SLA and polycom

[Carlos Rojas]

Hello, every body

Anyone set up, the sla sharing line appearances, in asterisk, I'm setting,
tha but, don't, work,  I change the sla.conf, extensions.conf, and sip.cfg,
but don't work fine.

Any one, could setup, tha?


Regards

Carlos Rojas
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Re: [asterisk-users] Beginner Question: Remote access

[Carlos Rojas]

Hello,

I use no-ip service, is similar than dyndns.com

Best Regards

asterisk-l...@puzzled.xs4all.nl wrote:

 On 09/07/2011 02:17 AM, A Dunor wrote:

 Hello list, I am a beginner at asterisk. I want to access my asterisk
 box from my laptop, on a different network (mobile hotspot). The
 asterisk box doesn't have a static ip, how do I connect with it using
 ssh or other such programs?

 Thanks for your guidance guys. It is highly appreciated.


 Use something like the dyndns service (dyn.com) via for example the
 ez-ipupdate client on the asterisk box and then an ssh client on your laptop
 which connects to the ssh server on your Asterisk box. Most secure is to use
 keys between the laptop and Asterisk server and avoid username/password
 authentication.

 Regards,
 Patrick


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Re: [asterisk-users] how to know how many calls are on hold

[Carlos Rojas]

Can you send the logs in cli console for help you?


Regards

On Tue, May 17, 2011 at 9:16 AM, virendra ban hati virbh...@gmail.comwrote:

 hi list,

 please help me how to know how many calls are on hold.

 --



 -
 Thanks and regards

  Virendra Bhati
 +91-9172341457
 Asterisk Engineer


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Re: [asterisk-users] OUTBOUND CALLER ID

[Carlos Rojas]

Hello

Do you set your callerid in the context outgoing?

[outgoing]

exten = _X.,1,Set(CALLERID(num)=4663000)
exten = _X.,n,Dial(..

On Mon, May 9, 2011 at 4:45 AM, mahesh katta maheshka...@flexydial.comwrote:

 Sir ,

 this is not working


 On Mon, May 9, 2011 at 1:52 PM, A J Stiles 
 asterisk_l...@earthshod.co.ukwrote:

 On Monday 09 May 2011, mahesh katta wrote:
  Hi,
  THIS IS IN DUBAI.
 
  I am having PRI line with 100 DID's (00-99) and when we call to any
  landline or mobile number then it shows us our board number or pilot
 number
  (i.e 4663000 means 00)..

 In the context through which outgoing calls are placed, you need a step
 which
 sets the caller ID number.  For instance, part of our dialplan maps
 external
 phone numbers with the local part 707060 to 707072 to internal extensions
 301
 to 312 respectively.  Our E1 provider also requires us to include the STD
 code, minus the leading zero, for the town we are in -- and will silently
 anonymise the call if we try to send a caller ID that does not belong to
 us.

 So for outgoing calls, we have something like

 [ts-outgoing]
 exten = _0., 1, Set(localno=7070$[${CALLERID(num)}-240])
 exten = _0., 2, Set(CALLERID(num)=${STD}${localno})


 --
 AJS

 Answers come *after* questions.

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 --
 Best Regards,

 Mahesh Katta
 *BUZZ**WORKS* Business Services Private Limited
 BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
 (E) Mumbai 400069
 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
 Web http://www.buzzworks.com


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Re: [asterisk-users] I look ARI (Asterisk Recording Interface)

[Carlos Rojas]

Hello,

I use cri

http://www.tikalnetworks.com/voip/index.php?cid=38


Best regards

On Thu, Jun 24, 2010 at 3:22 AM, Mickael Monsieur 
mickael.monsi...@gmail.com wrote:

 Hello Bruce,

 This module is not reliable on FreePBX?
 You know if there is a open source web-voicemail for Asterisk?

 Best regards,
 Mickael.

 2010/6/23 bruce bruce bruceb...@gmail.com

 It's one of the bad modules that goes with FreePBX anyhow. The moment you
 go over 3000 recordings you are already in trouble. It's about time someone
 come up with a better moduel.

 On Wed, Jun 23, 2010 at 11:05 AM, Mickael Monsieur 
 mickael.monsi...@gmail.com wrote:

 Hello,
 I look ARI (Asterisk Recording Interface)
 the publisher site is closed...

 http://www.littlejohnconsulting.com/ari

 Thank you,
 Mickael

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Re: [asterisk-users] Individual PIN Code per Extension

[Carlos Rojas]

Hello,


I use Authenticate command in dialplan.


Regards

Carlos Rojas

On Wed, Aug 19, 2009 at 6:33 AM, James Mutuku listmut...@gmail.com wrote:

 Hellos,

 I have astersist 1.2 working with freepbx. I want to tie pin codes to
 extensions(users). How do I do this?

 --
 Best Regards,
 James Mutuku Ndeti
 Agile Systems Limited
 +254722490994
 www.agile.co.ke
 mutuku.wordpress.com

 Has your organization implemented a customer relationship management
 (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM
 can help you achieve better customer satisfaction and sales

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Re: [asterisk-users] multiple call dialing and playback an message

[Carlos Rojas]

Hello,

You need configure a queue, with agents for that.


Regards.



On Thu, Aug 20, 2009 at 11:22 AM, kaustuva...@bbsr.syscomes.com wrote:

 I have tried a lot like as
 exten = 123,1,Dial(SIP/114SIP/113SIP/115)

 and all the channels are dialing and if i answered any 3 of one, all the
 channels except which one i answered are hung up..

 I need all 3 channels are ringing and playback a message to any one or
 more.
 So how to do it???

 Please, help me as i am new asterisk user

 Thanks in Advance..


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Re: [asterisk-users] no ring tone

[Carlos Rojas]

Hello,

I never use externhost

y use \

externip=public ip

And work fine


Regards

On Fri, Aug 14, 2009 at 9:18 AM, Ott Rose sixfourimp...@hotmail.com wrote:

  how do i troubleshoot no ring tone. It was working and all i added was the
 lines below now it doesn't ring.

   Edit sip_nat.conf for proper NAT:
 localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your
 external hostname name here)
 externrefresh=10
 fromdomain=DOMAIN.com (Set your external domain name here)
 nat=yes
 qualify=yes
 canreinvite=no


   Add extra codecs to /etc/asterisk/sip_custom.conf
 allow=gsm allow=h261
 allow=h263
 allow=h263p
 videosupport=yes
 --
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 out.http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009

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Re: [asterisk-users] no ring tone

[Carlos Rojas]

Hello

One question

In sip.con or sip_additionals.conf, in freepbx, the context of your client
do you put
nat = yes

externip = 

You put your public ip.

Are you sure that?


Regards

On Fri, Aug 14, 2009 at 10:43 AM, Ott Rose sixfourimp...@hotmail.comwrote:

  i changed it and still didn't ring. however it did ring on one call to a
 cell phone but it hasn't done it again.

 --
 Date: Fri, 14 Aug 2009 09:39:33 -0500
 From: crt.ro...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] no ring tone

 Hello,

 I never use externhost

 y use \

 externip=public ip

 And work fine


 Regards

 On Fri, Aug 14, 2009 at 9:18 AM, Ott Rose sixfourimp...@hotmail.comwrote:

  how do i troubleshoot no ring tone. It was working and all i added was the
 lines below now it doesn't ring.

   Edit sip_nat.conf for proper NAT:
 localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your
 external hostname name here)
 externrefresh=10
 fromdomain=DOMAIN.com (Set your external domain name here)
 nat=yes
 qualify=yes
 canreinvite=no


   Add extra codecs to /etc/asterisk/sip_custom.conf
 allow=gsm allow=h261
 allow=h263
 allow=h263p
 videosupport=yes
 --
 Windows Live™: Keep your life in sync. Check it 
 out.http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009

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[asterisk-users] Help for Alcatel asterisk

[Carlos Rojas]

Hello everybody

I have an asterisk with an integration of alcatel pbx, by sip trunk, all
calls are fine, but tha calls calls that originate from a analog line,
the recipient is not listening, and that if they hear the call originates,
the lines are E1 in alcatel pbx.

When a asteris user call to analog line the call is ok.


Everyone, has been that problem?

I change asterisk version 1.4.21 to 1.4.18 but the same problem.

I saw  the cli

[Aug 12 16:15:40] WARNING[2997]: chan_sip.c:3927 sip_indicate: Don't know
how to indicate condition 9
[Aug 12 16:15:40] WARNING[2997]: channel.c:2369 ast_indicate_data: Unable to
handle indication 9 for 'SIP/4001-0a16f5c0'

Anyone can help me..


Regards
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Re: [asterisk-users] Fwd: User Authentication in sip.conf

[Carlos Rojas]

Hello,


In your sip.conf

You need

host=sip.xxx.com

or IP

don't work with dynamic


Regards

On Wed, Aug 12, 2009 at 8:27 AM, harry R rhm.noa...@gmail.com wrote:

 Dear all,
  I want to setup the incoming calls, that don't use authentication in
 sip.conf file.
  My configurations as follows,

 [2000]
 type=peer
 host=dynamic
 insecure=port,invite; (both)
 context=Testing

 But when I call '2000', I noticed the following message in Asterisk
 console,

 NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to
 authenticate user Velusamy sip:7...@192.168.1.222sip%3a...@192.168.1.222
 ;tag=yj66acQcycvrN


 Hi

 I'm not sure about this but I think that it may cause by a bad setting in
 your softphone or your VOIP phone.
 Velusamy is a terminal that you have configured in sip.conf ?


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Re: [asterisk-users] Voicemail attachments not working

[Carlos Rojas]

Hello,

Your smtp server is on?


Best regards


Carlos Rojas

On Mon, Jul 6, 2009 at 7:30 PM, Steve Anness steve.ann...@gmail.com wrote:

  Today I discovered that voicemail attachments are not working on our
 latest asterisk server (version 1.4.24.1).  I have two other asterisk
 servers that I maintain but I didn’t do the configuration on these so this
 is my first time that I have done the voicemail.conf.   I get an email but
 there is no attachment.  Maybe there is something else I need to configure
 that I don’t know about?  Here is my actual config, the only difference is I
 removed all the mailboxes for the purpose of sharing with the world.
  However, I have made sure there are not spaces between fields as I hear
 that causes problems.

 [general]

 format = gsm|wav49|wav
 attach = yes
 serveremail = asterisk
 serveremail = nore...@mustangintl.com
 mailcmd = /usr/sbin/sendmail -v -t -f aster...@hisg-it.net
 maxlogins = 3
 emaildateformat = %A, %B %d, %Y at %r
 sendvoicemail = yes  ; Allow the user to compose and send a voicemail while
 inside
 emailsubject = [PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}

 [zonemessages]
 eastern = America/New_York|'vm-received' Q 'digits/at' IMp
 central = America/Chicago|'vm-received' Q 'digits/at' IMp
 central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours'
 military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
 european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM

 [default]
 116 = 1149,employee,emplo...@domain.org


 Suggestions?

 Thank you everyone in advance.

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Re: [asterisk-users] Trunk SIP and configuration

[Carlos Rojas]

Hello,

I don't speak english very well but i think.


[operador]
qualify=yes
nat=yes
host=192.168.700.50
insecure=invite,port
canreinvite=no
context=default
disallow=all
allow=ulaw
allow=g729

in your extensions.conf

exten = _00X,1, Dial (SIP/operador/${EXTEN},60,tT)



Best Regards


Carlos Rojas

On Wed, Apr 1, 2009 at 10:45 AM, ludo perrot ludoper...@gmail.com wrote:

 hello,

 I am beginning to asterisk.
 I have a sip trunk access to operator and VPN access with operator.
 i booked 10 sda numbers.

 IP adress asterisk : 192.168.600.1
 IP adress operator : 192.168.700.50
 i can ping on 192.168.700.50


 # cat sip.conf
 [general]
 context=default
 srvlookup=yes
 port = 5060
 disallow=all
 allow=gsm
 allow=alaw
 allow=ulaw

 [1000]
 username=1000
 type=friend
 qualify=yes
 secret=3615
 nat=no
 host=192.168.600.3
 canreinvite=no
 context=appels_entrants

 [Catherine]
 usename=1010
 type=friend
 qualify=yes
 secret=5768
 nat=yes
 host=192.168.600.4
 canreinvite=yes
 context=default
 disallow=all
 allow=ulaw

 # extensions.conf
 exten = _00X,1, Dial (SIP/192.168.700.50/${EXTEN})

 How do I configure IP operator ?
 I have 10 numbers sda. Where do I configure sda numbers ?

 Thanks.
 Ludovic




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Re: [asterisk-users] having problems with asterisk

[Carlos Rojas]

Hello

asterisk   -vvvgc


Regards

On Wed, Dec 10, 2008 at 7:45 PM, Scott Berry n7...@northlc.com wrote:

  Hello there,

 I am reading Asterisk: The Future of Telephony Chapter four.  I am using a
 Ubuntu box with Asterisk precompiled at this time so I can learn.  I am
 finding that I am having a problem when I do asterisk -r from the command
 line.  It says:
 Unable to connect remotely (are you sure that
 /var/run/asterisk/asterisk.ctl is available.)  The answer to this question
 is yes.  I also see through my logs that there are over a hundred modules
 loading and I just want the timing interface at this time.  I do not have
 hardware to use but I set up Asterisk as the boo recommends in Chapter
 four.  Can anyone help me in the proper direction.

 1.  I don't need all one hundred modules I just want the timing interface.

 2.  I don't see why asterisk -r is not working.

 Thanks for your help and included is my messages file.




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Re: [asterisk-users] canreinvite=yes problem

[Carlos Rojas]

Hello,

canreinvite, don't work with all softphone or hardphone.


Regards

On Wed, Dec 3, 2008 at 12:38 PM, BERGANZ François 
[EMAIL PROTECTED] wrote:

  Someone have a solution for me ?



 *De :* [EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] *De la part de* BERGANZ François
 *Envoyé :* mercredi 3 décembre 2008 18:24
 *À :* asterisk-users@lists.digium.com
 *Objet :* [asterisk-users] canreinvite=yes problem





 Hello,



 I need to test canreinvite=yes with 2softphones and 1 asterisk.



 I want to have that :
 http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outbridge.png

 But I have that http://www.zimagez.com/zimage/canreinvite.php



 Canreinvite=yes work for all phones or just asterisk?...



 Can you help me?



 Thank you

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Re: [asterisk-users] ztd-ethmf

[Carlos Rojas]

Hello,

Do you download zaptel of Redfone website?

Best Regards

On Fri, Aug 22, 2008 at 6:28 PM, Bill Michaelson [EMAIL PROTECTED] wrote:

 I expected to find th module ztd-ethmf[.c...] in support of the redfone
 TDMoE product in my zaptel distro (I have 1.4.11).  But it's not there.  I
 am awaiting a response to a trouble ticket from redfone.  Can anyone give me
 a jumpstart?  I can't seem to google this up.

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Re: [asterisk-users] Grandstream

[Carlos Rojas]

Hello,

Do you redirected the rtp ports to your phone?

usually 1 - 2  defautl rtp ports


Best Regards


Carlos Rojas

On Thu, May 22, 2008 at 8:48 AM, Phibee Network Operation Center 
[EMAIL PROTECTED] wrote:

 I have a problem connecting a Grandstream ipphone to an asterisk.

 The ipphone is behind a nat router, I redirected UDP 5060 and 5004 to my
 phone.
 It connects well to the asterisk server. I can call outside and receive
 calls from outside without any problems.

 But if I call from this ipphone to another ipphone connected on the same
 asterisk server, using internal dialing, I can hear my correspondant, but
 he
 cannot.

 Do you have any idea?
 Thanks for advance.


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Re: [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help

[Carlos Rojas]

Hello,

Do your verify, the codecs, of both clients, in your sip.conf?

What codec do you use?

Best Regards

On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote:

 Hi,
 I am sorry my questinos are too fundamental.  I am new to Asterisk, and
 hope to catch up as fast as I can.

 Problem 1:

 I have my SIP  client ( in one PC .102) and SIP server ( in another PC
 .101) within the same land.  They can make SIP connection, but when the SIP
 client makes call to play an audio file, I can only hear a beat sounds,
 and then nothing else.  In the console, I can see:
 *CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2001-081dd6e0, )
 in new stack
 -- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/2001-081dd6e0, 2000)
 in new stack
 Sent RTP packet to  58.251.75.228:9956 (type 00, seq 037718, ts
 000160, len 000160)
 -- SIP/2001-081dd6e0 Playing 'vm-intro' (language 'en')
 Sent RTP packet to  58.251.75.228:9956 (type 00, seq 037719, ts
 000320, len 000160)
 Sent RTP packet to  58.251.75.228:9956 (type 00, seq 037720, ts
 000480, len 000160)
 Sent RTP packet to  58.251.75.228:9956 (type 00, seq 037721, ts
 000640, len 000160)
 Got  RTP packet from192.168.1.102:8000 (type 00, seq 06, ts
 1373137124, len 000160)
 Sent RTP packet to  192.168.1.102:8000 (type 00, seq 037722, ts
 000800, len 000160)
 Sent RTP packet to  192.168.1.102:8000 (type 00, seq 037723, ts
 000960, len 000160)

 Is it the prolem?  First it sends to the public address of the the router,
 then it sends to the virtual IP.  Is this the problem that causing my to
 hear just one beat sound and then no audio?

 Problem 2:

 The problem is isolated from Problem 1, cuz I run the SIP client on the
 same machine as the server, so there should not be network problem.  I
 recorded some voice mails and they are stored as .wav files ok.  When I
 tried to hear back the message, It does not work.  Is there any
 configuration that I have to go through to have Asterisk to play .wav file?


 Thank you very much in advance for all your kind help.

 Pete


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[asterisk-users] asterisk gateway

[Carlos Rojas]

Hello everybody

Anyone, to know a gateway that works with nextel simm cards?
I'm looking for them, in internet, but I did'n look.

Best regards
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Re: [asterisk-users] asterisk on Hp servers

[Carlos Rojas]

Hello,

Remember, that linux has problems with irq and pci cards of digium, do you
have 3 digium card, and don't have any problems ?


Best Regards

On Jan 5, 2008 11:01 PM, Eric S López [EMAIL PROTECTED] wrote:

  Gres,

 Me, as an asterisk and linux newbie installed redhat 4 (without the gui)
 on a Proliant HP server with 3 digium cards, had no problems with the
 installation and it is running without problems for 18+ months now.  You
 shouldn't have any problems provided that your linux distro has all the
 prerequisite packages mentioned on the asterisk guides, if not you will have
 to install them prior to the asterisk install.

 If you are using PCI cards with your asterisk, don't forget to check that
 the voltage of the cards matches the voltage on the motherboard.

 Best regards and good luck,

 Eric ... from Guatemala.

 - Original Message -
 *From:* Gres + [EMAIL PROTECTED]
 *To:* asterisk-users@lists.digium.com
 *Sent:* Saturday, January 05, 2008 1:50 PM
 *Subject:* [asterisk-users] asterisk on Hp servers


 please can anyone help me knowing if i can install Linux and Asterisk on
 HP servers
 **


 --
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 Start
 now!http://www.microsoft.com/windows/shop/specialoffers.mspx?ocid=TXT_TAGLM_CPC_MediaCtr_bigscreen_012008


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[asterisk-users] Softswitch digim

[Carlos Rojas]

Hello averybody,


I'm looking the softswitch in digium website, anyone test the softswitch?


Best Regards
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Re: [asterisk-users] Copy or Make + Make Install

[Carlos Rojas]

Hello,

Only copy the configuration files, extensions.conf, sip.conf, iax.conf
,

Best regards



On Nov 27, 2007 1:27 PM, bilal ghayyad [EMAIL PROTECTED] wrote:

 Hi List;

 If I have a running Asterisk on one machine and I need
 to have another Asterisk on another machine, can I
 copy the files from the first running Asterisk machine
 to the new machine or I have to do the ./configure +
 make + make install?

 If I can copy, then which directories (and files) need
 to be copied?

 What if my new machine have other kernel version that
 first machine?

 Regards
 Bilal



  
 
 Never miss a thing.  Make Yahoo your home page.
 http://www.yahoo.com/r/hs

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Re: [asterisk-users] TDM400P not answering or making calls

[Carlos Rojas]

Heloo,

I think that your error is:

zaptel.conf:

---
fxsks=1
loadzone= uk
defaultzone = uk

zapata.conf:

[channels]
language=en
context=incoming
signalling=fxs_ks
busydetect=yes
busycount=4
callprogress=no
relaxdtmf=yes
callwaiting=no
callwaitingcallerid=no
threewaycalling=no
transfer=yes
cancallforward=yes
usecallerid=no
cidsignalling=v23
cidstart=polarity
callerid=no
hidecallerid=no
echotraining=yes
echocancel=yes
echocancelwhenbridged=yes
musiconhold=default
immediate=no

channel = 3


Best Regards


On 9/11/07, Tom Playford [EMAIL PROTECTED] wrote:

 Hello,

 I have recently purchased a TDM400P card with one FXO expansion card,
 and I'm having problems.

 The card does not pick up incoming calls. Asterisk detects the ringing
 line and rings various SIP phones as required. When a sip phone
 answers, the sip user hears nothing and the PSTN user continues to
 hear ringing.  Here is the asterisk output for an incoming call:

 -
   == Starting post polarity CID detection on channel 3
 -- Starting simple switch on 'Zap/3-1'
 -- Executing Set(Zap/3-1, CALLERID(all)=call to 322817) in new
 stack
 -- Executing Dial(Zap/3-1, Local/[EMAIL PROTECTED]|45) in new stack
 -- Called [EMAIL PROTECTED]
 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/203|30) in new
 stack
 -- Called 201
 -- SIP/203-0814c448 is ringing
 -- Local/[EMAIL PROTECTED],1 is ringing
 -- SIP/201-081445e8 is ringing
 -- SIP/201-081445e8 answered Local/[EMAIL PROTECTED],2
 -- Local/[EMAIL PROTECTED],1 stopped sounds
 -- Local/[EMAIL PROTECTED],1 answered Zap/3-1
   == Spawn extension (special, 601, 1) exited non-zero on
 'Local/[EMAIL PROTECTED],2'
   == Spawn extension (special, 601, 1) exited non-zero on
 'Local/[EMAIL PROTECTED],2'
 Sep 11 19:25:55 WARNING[3073]: chan_zap.c:3934 zt_handle_event:
 Ring/Off-hook in strange state 6 on channel 3
   == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/3-1'
 -- Hungup 'Zap/3-1'
 -

 I have set opermode to 'UK' (as I'm in the UK), and dmesg confirms the
 setting.

 Outgoing calls also fail, the SIP user hears nothing, yet asterisk
 claims that the call has been picked up. Here is the asterisk log,
 looks perfectly normal:

 -
 -- Executing Dial(SIP/201-08154a38, Zap/3/0800800800) in new stack
 -- Called 3/0800800800
 -- Zap/3-1 answered SIP/201-08154a38
 -

 I am running Debian Etch with kernel 2.6.18 and asterisk version 1.2.13.

 I'm beginning to think it's a fault with the expansion card... anyone
 else got any ideas? Oh and the BT line is fine, it works (as well as
 can be expected) with a X100P card I have.


 Thanks,

 Tom

 Attached:

 dmesg output:
 
 apata Telephony Interface Registered on major 196
 Zaptel Version: 1.2.16
 Zaptel Echo Canceller: MG2
 ACPI: PCI Interrupt :00:0b.0[A] - Link [LNKD] - GSI 12 (level,
 low) - IRQ 12
 Freshmaker version: 73
 Freshmaker passed register test
 Module 0: Not installed
 Module 1: Not installed
 Module 2: Installed -- AUTO FXO (UK mode)
 Module 3: Not installed
 Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules)
 Registered tone zone 4 (United Kingdom)
 -

 zapata.conf:
 
 [channels]
 language=en
 context=incoming
 signalling=fxs_ks
 busydetect=yes
 busycount=4
 callprogress=no
 relaxdtmf=yes
 callwaiting=no
 callwaitingcallerid=no
 threewaycalling=no
 transfer=yes
 cancallforward=yes
 usecallerid=no
 cidsignalling=v23
 cidstart=polarity
 callerid=no
 hidecallerid=no
 echotraining=yes
 echocancel=yes
 echocancelwhenbridged=yes
 musiconhold=default
 immediate=no
 -

 zaptel.conf:

 ---
 fxsks=3
 loadzone= uk
 defaultzone = uk
 ---

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Re: [asterisk-users] Locating Asterisk documentation after installation

[Carlos Rojas]

Hello,
Do you have install  doxygen?

Best regards

On 8/13/07, Philipp Kempgen [EMAIL PROTECTED] wrote:

 MOSBAH ABDELKADER wrote:

  After installing Asterisk, i have installed the docs by make progdocs.
 
  But i don't know where to locate this documentation.

 Maybe /usr/src/asterisk-*/doc/api/ ?


 Regards,
   Philipp Kempgen

 --
 amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? - http://www.das-asterisk-buch.de
   My pick of the month: rfc 2822 3.6.5

 Geschäftsführer: Stefan Wintermeyer
 Handelsregister: Neuwied B 14998

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Re: [asterisk-users] Problems building zaptel 1.4.4

[Carlos Rojas]

Hello,

In Asterisk 1.4 and zaptel 1.4,

don't work make linux26,

zaptel and asterisk works with kernel 26, and only work with


./configure
make menuselect
make
make install

Best Regards

Carlos Rojas
Lima - Peru

On 7/31/07, hugolivude [EMAIL PROTECTED] wrote:

 Hi,

 I'm having trouble compiling zaptel 1.4.4 on SUSE 10.1.  I'm really
 only interested in getting ztdummy to work because this is a dev
 machine with no zaptel h/w.  SUSE 10.1 is a 2.6 kernel:

 asterisk-dev:/home/hugh # uname -r
 2.6.16.13-4-default

 It seems that my problem is related to autoconf.h - I cannot find that
 file:

 asterisk-dev:/home/hugh #  find / -name 'autoconf.h'

 comes up empty.

 As a result make linux26 doesn't work:

 asterisk-dev:/usr/src/packages/SOURCES/Asterisk/zaptel-1.4.4 # make
 linux26
 grep: /lib/modules/2.6.16.13-4-default/build/include/linux/autoconf.h:
 No such file or directory
 make: *** No rule to make target `linux26'.  Stop.
 asterisk-dev:/usr/src/packages/SOURCES/Asterisk/zaptel-1.4.4 #

 I've noticed that the installs come with a menuselect feature now.  I
 had to run make menuselect twice, but it seemed to complete OK.  I
 fooled aound a little inside menuselect delescting everything but
 ztdummy, but still no joy with autoconf.h missing:

 asterisk-dev:/usr/src/packages/SOURCES/Asterisk/zaptel-1.4.4 # make
 grep: /lib/modules/2.6.16.13-4-default/build/include/linux/autoconf.h:
 No such file or directory
 grep: /lib/modules/2.6.16.13-4-default/build/include/linux/autoconf.h:
 No such file or directory
 make[1]: Entering directory `/usr/src/packages/SOURCES/Asterisk/zaptel-
 1.4.4'
 gcc gendigits.c  -lm -o gendigits
 ./gendigits  tones.h
 gcc -o makefw makefw.c
 ./makefw tormenta2.rbt tor2fw  tor2fw.h
 Loaded 69900 bytes from file
 ./makefw pciradio.rbt radfw  radfw.h
 Loaded 42096 bytes from file
 make -C /lib/modules/2.6.16.13-4-default/build
 SUBDIRS=/usr/src/packages/SOURCES/Asterisk/zaptel-1.4.4 modules
 make[2]: Entering directory `/usr/src/linux-2.6.16.13-4-obj/i386/default'
 make[2]: *** No rule to make target `modules'.  Stop.
 make[2]: Leaving directory `/usr/src/linux-2.6.16.13-4-obj/i386/default'
 make[1]: *** [modules] Error 2
 make[1]: Leaving directory `/usr/src/packages/SOURCES/Asterisk/zaptel-
 1.4.4'
 make: *** [all] Error 2

 Any ideas that could help me out?

 Thanks,
 Hugh

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Re: [asterisk-users] asterisk or asterisknow

[Carlos Rojas]

Hello,

I prefere, asterisk

Best Regards


On 7/31/07, Al lists [EMAIL PROTECTED] wrote:

 You can use both Asterisk or AsteriskNow to have meetme (conference room)

 On 7/30/07, fateme fatah [EMAIL PROTECTED]  wrote:

  Hi:
  I want to have conference call service.You offer  me use asterisk or
  asterisknow.
  Regards.
 
  --
  Be a better Globetrotter. Get better travel answers
  http://us.rd.yahoo.com/evt=48254/*http://answers.yahoo.com/dir/_ylc=X3oDMTI5MGx2aThyBF9TAzIxMTU1MDAzNTIEX3MDMzk2NTQ1MTAzBHNlYwNCQUJwaWxsYXJfTklfMzYwBHNsawNQcm9kdWN0X3F1ZXN0aW9uX3BhZ2U-?link=listsid=396545469from
  someone who knows.
  Yahoo! Answers - Check it out.
 
 
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Re: [asterisk-users] Calling to users in other asterisk servers

[Carlos Rojas]

Hello,

in your sip.conf do you have

[yourprovider]
username=
fromuser=
secret=
host=another.server.com
nat=yes
.
.
.
.

and in your extensions.conf

And the extensions.conf:
...
exten = _X.,1,Dial,SIP/yourprovider

...

Best Regards

sip:[EMAIL PROTECTED] )

On 7/29/07, Ary Junior [EMAIL PROTECTED] wrote:

 Ok, my firewall port forward rules:

 TCP5004 - 5082192.168.254. 2 UDP5004 - 5082192.168.254. 2 TCP4569
 192.168.254. 2UDP 4569192.168.254. 2UDP1 - 2192.168.254 . 2
 And it dont works... Any configuration in special for make call the to
 users in another asterisk servers?

 Thanks very much!!!

 On 7/28/07, Carlos Rojas [EMAIL PROTECTED] wrote:
 
  Hello,
 
  Do you have porf forwardin for SIP protocol in your firewall?
 
  SIP:  5060  udp
 
  rtp  1 - 2 udp (default)
 
  and IAX2 4569  udp
 
 
  Best Regards
 
 
  Carlos Rojas
 
  On 7/28/07, Ary Junior [EMAIL PROTECTED]  wrote:
  
   Hi, Im a asterisk newbie and I've configured an asterisk server here
   in my house... in my LAN two users can login and call to each other, but
   when I try to call an user in another asterisk server outside my LAN (
   sip:[EMAIL PROTECTED] ) it dont work... if the person outside is
   conected on my server it works fine... My asterisk server is behind a
   firewall and portfowarding... it is possible?
  
   Thanks very much!!!
  
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Re: [asterisk-users] Calling to users in other asterisk servers

[Carlos Rojas]

Hello,

Do you have porf forwardin for SIP protocol in your firewall?

SIP:  5060  udp

rtp  1 - 2 udp (default)

and IAX2 4569  udp


Best Regards


Carlos Rojas

On 7/28/07, Ary Junior [EMAIL PROTECTED] wrote:

 Hi, Im a asterisk newbie and I've configured an asterisk server here in my
 house... in my LAN two users can login and call to each other, but when I
 try to call an user in another asterisk server outside my LAN (
 sip:[EMAIL PROTECTED] ) it dont work... if the person outside is
 conected on my server it works fine... My asterisk server is behind a
 firewall and portfowarding... it is possible?

 Thanks very much!!!

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Re: [asterisk-users] Asterisk and ATA-186 question-- calling one port from the other port..

[Carlos Rojas]

Hello,

I
Check this page:
http://www.asterisk.net.au/general/1/

It's very interesting


Best Regards

Carlos Rojas
On 7/18/07, Dmytro Mishchenko [EMAIL PROTECTED] wrote:


Tim Reimers wrote:


 Hi -

 I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both
ports.

 I need to be able to call one port from the other-- the idea is to have
 two phones in two different locations that _can_ call each other.

 So, in reading the Asterisk Wiki and other sites, the best documentation
 I found was this:




 *http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt*
 **
 ***Note that each line must have it's own distinct and complete
 configuration, and if you use both lines on the ATA-186, it will
 REGISTER twice.  Further note that you cannot call one line from the
 other on the same device using the direct extension numbers, so you
 will have to be clever about naming and aliases within Asterisk.  That
 is outside the scope of this document.
 *
 However, that _specifically_ says that in the provided config, you
 cannot call one port from the other port using direct lines--
 It does indicate that you CAN in fact work that out, using naming and
 aliases within Asterisk.

 Therefore, I assume that it IS possible to use an ATA like this---
 but that the author of this particular doc either doesn't know how (but
 does know it can be done)
 or just didn't want to go into it in a low-level howto.

 So ---

 Does anyone know how to do this?


Check this page:
http://www.voip-info.org/wiki/view/Cisco+ATA+186+SIP+and+Asterisk+-+HowTo

There is example for configuring two lines.

Dmitry.



 thanks, Tim



 Most days, there are several fires burning at once. Some days, what's
 burning is your fire extinguisher.
 To err is human; to truly screw it up requires the root password.






 

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Re: [asterisk-users] open source screen pop software for asterisk

[Carlos Rojas]

Hi,

I work with

gnudialer
vicidal

Best Regards


On 7/14/07, Todd H [EMAIL PROTECTED] wrote:


I like ADM as it has a URL popup feature (open a URL with a DID or
CallerID in URL).  The problem is that for each call, I tend to get 4
or 5 popups... But as the other author said, there are many
programs to choose from...
   Todd

On Jul 13, 2007, at 11:54 PM, RENZZO SOTOMAYOR wrote:

 Hi! I am new here. Well I'm doing a call center using asterisk and
 I'm looking for an open source screen pop software to pop the
 caller's information, its call history  and others things. i was
 looking around and find the U-rang2 the problem is that it isn't
 open source. if someone knows about an open source screen pop
 please tell me.

 thanks in advance

 renzzo


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Re: [asterisk-users] simple dial plan question

[Carlos Rojas]

Hello,

In your sip.conf you don't have the user for you provider:

[yourprovider]
username=1234
secret=sdfdsf
host=sip.yourprovider.com
type=peer
...

In yor extensions.conf

[mycontext]

exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,103,Hangup

exten = 2001,1,Dial(SIP/2001,20)
exten = 2001,2,Hangup

exten = _001X.,1,Dial(SIP/[EMAIL PROTECTED],25)
exten = _001X.,2,Hungup()

For call out US, for example

Best Regards
On 6/18/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:


Dear asterisk users,

I need some help , I'm a little new  in VoIP , asterisk. I have
downloaded, compiled , installed. I make a simple configuration (I'm sorry
write the configuration)
1. sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
allow=all
context=default
register = user:[EMAIL PROTECTED]/2000

[2000]
type=friend
username=2000
secret=secret2000
host=dynamic
context=mycontex
maibox=2001

[2001]
type=friend
username=2001
secret=secret2001
host=dynamic
context=mycontext
maibox=2001

2.extensions.conf
[general]
statis=yes
writeprotect=yes

[default]
exten = _.,1,Congestion

[mycontext]

exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,103,Hangup

exten = 2001,1,Dial(SIP/2001,20)
exten = 2001,2,Hangup

So far so good.Well the asterisk it's working , 2000 can call 2001 and
2001 cam call 2000 usind a VoiP ATA adapter or a softphone.Well my
question is: I WANT TO WRITE A DIAL PLAN FOR USERS 2000 AND 2001 TO CALL A
NATIONAL NUMBER BY MY SIP PROVIDER sip.myvoipprovider.com (what do I have
to write in extensions.conf anf how)
It's good a combination , like by pressing 0 , both users have a dial tone
for outside, if is to much

Thanks a lot

eng. Alexandru Achim
National Institute Lasers Physics
Quantum Solid States Laboratory
Magurele,Bucharest,Romania


Sincerely




*
Acest email a fost verificat de catre NOD32 Antivirus

Serviciu oferit de catre ITSISTEM SERVICES SRL
Tel: 0752.304326 , 0752.304327
[EMAIL PROTECTED] http://www.itsistem.ro

**
  part000.txt - is OK
http://www.eset.com

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Re: [asterisk-users] Connect two Asterisk boxes through IVR Menu

[Carlos Rojas]

Hello,

I take the example:

exten = 300,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN},30)


Best Regards


On 5/26/07, Alex Balashov [EMAIL PROTECTED] wrote:



Matt,

On Sat, 26 May 2007, Matt Darnell wrote:

 exten = _3xx,1,dial(IAX2/{$EXTEN})
 exten = 300,1,dial(IAX2/301)

   You do not appear to be specifying a destination host, i.e. the other
endpoint of the IAX trunk.  Asterisk does not have an automatic way of
resolving such remote endpoints or their constituent extensions, at least
not without a facility that specifically furnishes such resolution such as
DUNDi.

   For an extension whose destination is on the remote Asterisk server,
try something like:

exten = 300,1,Dial(IAX2/@remote_peer/301)

   Further explanation at:

  http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels


--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
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Re: [asterisk-users] TDM400P usada?

[Carlos Rojas]

Hey

Look

http://www.asterisk-es.org

Best Regards

On 5/5/07, Cesar Benjamin Garcia Martinez [EMAIL PROTECTED] wrote:


 Bueno, no esta en chile, yo no estoy en chile, pero hablo español, soy de
mexico, asi que en parte tienes razón, pero tb creo que deberías haber
puesto de donde eres.



*De:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *En nombre de *Rodrigo Mercado
*Enviado el:* sábado, 05 de mayo de 2007 12:38
*Para:* Asterisk Users Mailing List - Non-Commercial Discussion
*Asunto:* Re: [asterisk-users] TDM400P usada?



Chile.



No hay listas en español, y si lo enviè en español es justamente porque si
alguien no lo habla no puede estar en CHILE, de todas formas muchas gracias
por la amabilidad de traducir mi correo.



saludos,



bye bye



On 5/5/07, *Tom Rymes* [EMAIL PROTECTED] wrote:

On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote:

 Alguien tiene una TDM400P con modulo FXS usada a la venta ??,
 obviamente a precio de tarjeta usada...


 saludos,


 Rodrigo Mercado S.

For anyone who is not a Spanish speaker, Rodrigo is looking for a
used TDM400P card with FXS modules. He is expecting a price that
would correspond with a used card. (In other words, cheap)

Rodrigo:

1.) ¿Donde estás? ¿Cómo podria alguien dar un precio sin saber donde
tendria que mandarlo? ¿España? ¿Puerto Rico? ¿Argentina?
2.) Si no hablas Inglés, seria mejor buscar una lista de Asterisk en
Español, porque la mayoria de las personas aqui no hablen Español.

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Re: [asterisk-users] Re: Balancing interrupts.

[Carlos Rojas]

Hello

And  lspci -vb   ??

Regards

On 5/4/07, Daniel Pittman [EMAIL PROTECTED] wrote:


Steve Edwards [EMAIL PROTECTED] writes:

 I see the following on one of my new servers:

 -ts10::sedwards:~$ cat /proc/interrupts
 CPU0   CPU1   CPU2   CPU3
0:29790452988620   87780075   87779501IO-APIC-edge  timer

[...]

 225:4611916 681023   84732445   89903138
IO-APIC-level  wct4xxp
 NMI:  0  0  0  0
 LOC:  181534588  181534654  181534653  181534652
 ERR:  0
 MIS:  0

 -ts10::sedwards:~$ ps -e | grep bal
   2633 ?00:00:00 irqbalance

 Should I be concerned that cpu1 is servicing only 700,000 interrupts
 from my te410p while cpu3 is servicing almost 90,000,000?

 I thought this is what irqbalance was for...

Actually, what you *really* want (for performance reasons) is to have
one CPU handle *all* the interrupts and all the threads that talk to
hardware for that card, if possible.

Every time you move the IRQ to a different CPU you lose a bunch of
cycles reloading data from main memory into the L2 and L1 cache, cycles
that can't be used elsewhere.

Binding that interrupt to one specific CPU -- and your NIC to a
different CPU -- is generally a good idea.  If you can keep the threads
that handle those signals and the hardware on that same CPU you increase
efficiency a bit more.

Moving the IRQ has plenty of cost and isn't a great plan.  :)

Regards,
Daniel
--
Digital Infrastructure Solutions -- making IT simple, stable and secure
Phone: 0401 155 707email: [EMAIL PROTECTED]
 http://digital-infrastructure.com.au/

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Re: [asterisk-users] Yeastar Cards

[Carlos Rojas]

Hello

I'd like to know too



On 4/2/07, Gustavo Felisberto [EMAIL PROTECTED] wrote:


I am in the process of buying a TDM800 card from Yeastar (

http://www.yeastar.com/products.asp?TypeName=TDM800%20PCI%20CardcTypeName=1)
Any one has tested this cards? How reliable are them? I am specially
interested
in the FXO/FXS module.


--
Gustavo Felisberto
(HumpBack)
Web: http://dev.gentoo.org/~humpback
Blog: http://blog.felisberto.net/

It's most certainly GNU/Linux, not Linux. Read more at
http://www.gnu.org/gnu/why-gnu-linux.html .
-


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