Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

[SamyGo]

Ok Good. It always feel good to add +1 to the list of resolved user-list
issues.

As I'm not a POWERFUL guy, I can't simply ask my ISP to unblock it.


It takes a VPN or in near future WebRTC(in other words "Knowledge") to
become one powerful guy. With these technologies you don't need to care
what your ISP or govt. is blocking.

Where there is will, there are ways.

BR
Sammy



On Fri, Aug 10, 2012 at 8:09 PM, Sazzad  wrote:

> But still contact your ISP and get them to un-block your port 5060.  You
>> paid
>> them for an Internet connection; and an Internet connection means *all*
>> ports,
>> not just *some* ports.
>>
>
> There has been lots misuses of VOIP here and government policies are
> strange. Instead, lots of illegal VOIP businesses run. As I'm not a
> POWERFUL guy, I can't simply ask my ISP to unblock it.
>
> Law and constitution, were a means to protect us. (Now, don't smile. I
> know it amuses you too.)
>
> --
> Sincerely,
> Sazzad Bin Kamal
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Debian 7/Asterisk TLS bug and others

[Paul Belanger]

On 12-08-10 06:20 PM, Daniel Pocock wrote:



Debian is very conservative about accepting updates during the `freeze'
process - they will most likely want to see a 1.8.13.2 release with ONLY
the most essential fixes

a) is anyone else aware of these bugs?

b) what essential changes should go into 1.8.13.2 for Debian?


We don't need to release a 1.8.13.2 release of Asterisk.  Once the issue
has been fixed in the 1.8 release branch, it would just be back-ported
into a Debian patch for the package.


My impression was that a 1.8.13.2 release would be as conservative as
any patches back-ported for the Debian package.  It's not necessary, but
it might be a convenient way to achieve the same goal.

Is Digium officially endorsing 1.8.13 for wheezy in any way?

No. Digium nor the Asterisk Project has anything to do with the package 
within Debian.  In fact, most of the work is done by Tzafrir.



Is anyone officially working on this particular problem already?  I was
tempted to have a closer look at it, but don't want to duplicate an
effort that is already underway elsewhere.


Best to check JIRA and see.  Actually, does the issue even exist in JIRA?

--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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

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

Re: [asterisk-users] Debian 7/Asterisk TLS bug and others

[Daniel Pocock]

>> Debian is very conservative about accepting updates during the `freeze'
>> process - they will most likely want to see a 1.8.13.2 release with ONLY
>> the most essential fixes
>>
>> a) is anyone else aware of these bugs?
>>
>> b) what essential changes should go into 1.8.13.2 for Debian?
>>
> We don't need to release a 1.8.13.2 release of Asterisk.  Once the issue
> has been fixed in the 1.8 release branch, it would just be back-ported
> into a Debian patch for the package.

My impression was that a 1.8.13.2 release would be as conservative as
any patches back-ported for the Debian package.  It's not necessary, but
it might be a convenient way to achieve the same goal.

Is Digium officially endorsing 1.8.13 for wheezy in any way?

Is anyone officially working on this particular problem already?  I was
tempted to have a closer look at it, but don't want to duplicate an
effort that is already underway elsewhere.


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

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

Re: [asterisk-users] Debian 7/Asterisk TLS bug and others

[Paul Belanger]

On 12-08-10 04:47 PM, Daniel Pocock wrote:



Debian 7 is currently in the `freeze' status with 1.8.13 - that means
Debian 7 is very likely to release 1.8.13 and be carrying it for the
next 2-3 years (typical lifetime of a Debian release)

I run 1.8.8.  TLS has a bug: it fails to receive BYE over the TLS
connection from my Polycom phone.

I tried 1.8.13, the version in Debian 7, and found a more severe bug:
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=683956
The TLS clients can't connect at all, this looks like a really bad
regression from 1.8.8

I've looked at 1.8.(14, 15, 16-rc1) and their changelogs don't mention
any fix.

Debian is very conservative about accepting updates during the `freeze'
process - they will most likely want to see a 1.8.13.2 release with ONLY
the most essential fixes

a) is anyone else aware of these bugs?

b) what essential changes should go into 1.8.13.2 for Debian?

We don't need to release a 1.8.13.2 release of Asterisk.  Once the issue 
has been fixed in the 1.8 release branch, it would just be back-ported 
into a Debian patch for the package.


--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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

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

[asterisk-users] Asterisk 11.0.0-beta1 Now Available!

[Asterisk Development Team]

The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 11.0.0.  This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

All interested users of Asterisk are encouraged to participate in the
Asterisk 11 testing process.  Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira.  It is also very useful to see
successful test reports.  Please post those to the asterisk-dev mailing list.
All Asterisk users are invited to participate in the #asterisk-testing channel
on IRC to work together in testing the many parts of Asterisk.  

Asterisk 11 is the next major release series of Asterisk.  It will be a Long
Term Support (LTS) release, similar to Asterisk 1.8.  For more information about
support time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 11, please see the
Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

A short list of new features includes:

* A new channel driver named chan_motif has been added which provides support
  for Google Talk and Jingle in a single channel driver.  This new channel
  driver includes support for both audio and video, RFC2833 DTMF, all codecs
  supported by Asterisk, hold, unhold, and ringing notification. It is also
  compliant with the current Jingle specification, current Google Jingle
  specification, and the original Google Talk protocol.

* Support for the WebSocket transport for chan_sip.

* SIP peers can now be configured to support negotiation of ICE candidates.

* The app_page application now no longer depends on DAHDI or app_meetme. It
  has been re-architected to use app_confbridge internally.

* Hangup handlers can be attached to channels using the CHANNEL() function.
  Hangup handlers will run when the channel is hung up similar to the h
  extension; however, unlike an h extension, a hangup handler is associated with
  the actual channel and will execute anytime that channel is hung up,
  regardless of where it is in the dialplan.

* Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
  allows you to execute a dialplan subroutine on a channel before a call is
  placed but after the application performing a dial action is invoked. This
  means that the handlers are executed after the creation of the caller/callee
  channels, but before any actions have been taken to actually dial the callee
  channels.

* Log messages can now be easily associated with a certain call by looking at
  a new unique identifier, "Call Id".  Call ids are attached to log messages for
  just about any case where it can be determined that the message is related
  to a particular call.

* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
  Asterisk. Unlike traditional ACLs defined in specific module configuration
  files, Named ACLs can be shared across multiple modules.

* The Hangup Cause family of functions and dialplan applications allow for
  inspection of the hangup cause codes for each channel involved in a call.
  This allows a dialplan writer to determine, for each channel, who hung up and
  for what reason(s).

* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
  lets you set some of the configuration options from the general section
  of features.conf on a per-channel basis. FEATUREMAP() lets you customize
  the key sequence used to activate built-in features, such as blindxfer,
  and automon.

* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
  and callgroups to be defined for several channel drivers.

* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation

A full list of all new features can also be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/11/CHANGES

For a full list of changes in the current release, please see the ChangeLog.

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1

Thank you for your continued support of Asterisk!








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

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

[asterisk-users] Debian 7/Asterisk TLS bug and others

[Daniel Pocock]

Debian 7 is currently in the `freeze' status with 1.8.13 - that means
Debian 7 is very likely to release 1.8.13 and be carrying it for the
next 2-3 years (typical lifetime of a Debian release)

I run 1.8.8.  TLS has a bug: it fails to receive BYE over the TLS
connection from my Polycom phone.

I tried 1.8.13, the version in Debian 7, and found a more severe bug:
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=683956
The TLS clients can't connect at all, this looks like a really bad
regression from 1.8.8

I've looked at 1.8.(14, 15, 16-rc1) and their changelogs don't mention
any fix.

Debian is very conservative about accepting updates during the `freeze'
process - they will most likely want to see a 1.8.13.2 release with ONLY
the most essential fixes

a) is anyone else aware of these bugs?

b) what essential changes should go into 1.8.13.2 for Debian?


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

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

[asterisk-users] ConfBridge

[Jerry Geis]

I am starting to use ConfBridge and not MeetMe in asterisk 10.

I have everything converted over EXCEPT.

I am using an AGI and AMI to bring phones into a conf automatically.
When I do that the conf is going just fine - however - I head beep, 
beep, beep.


I have every sound listed in confbridge.conf turned off.

I cannot pinpoint the beep beep beep. It continues through the whole conf.

Is the because my AGI has exited that brought all the people into the 
conference?

How can I turn off the beep beep beep?

Thanks,

Jerry

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

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

Re: [asterisk-users] asterisk and meetme

[Matthew Jordan]

- Original Message - 

> From: "Matt Hamilton" 
> To: asterisk-users@lists.digium.com
> Sent: Friday, August 10, 2012 12:15:14 PM
> Subject: Re: [asterisk-users] asterisk and meetme

> > ConfBridge is the preferred conference application in Asterisk 10+.
> > While
> > MeetMe is currently deprecated, you can still enable it and run it
> > in
> > Asterisk 10+.

> What's going to happen to SLA (which is heavily integrated with
> MeetMe)? Will the functionality be ported to ConfBridge?

So, there was some discussion around that very problem.  The result of the
discussion is ASTERISK-20134 - which is that starting rather soon (in a future
version of Asterisk 10 that has not yet been put into RC status), app_meetme
will be put into the 'extended' support category for that very reason.

As far as porting the SLA functionality to use the internal Bridging API
that ConfBridge is based on, that sounds like a great idea - but it would
be a non trivial task.  We'd certainly welcome assistance from the community
in performing that migration.

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

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

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

Re: [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own

[Chad Wallace]

On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcus  wrote:

> I have two setups with SIP hardware phones as extensions and POTS
> lines as trunks. Internal SIP to SIP calls are crystal clear, but all
> calls bridged to POTS have a significant amount of static noise. The
> problem is that if I plug a POTS phone directly into the line, there
> is almost no static noise - the line is clean. It's like Asterisk (or
> the hardware) amplifies the static noise. What I've tried so far:
> 
> 1. Connect Asterisk with a short cable directly into the master phone 
> socket, where it enters the building.
> 2. One of the lines carries ADSL - so I double filtered it.
> 3. Tried three different phone sets (one Grandstream, two Cisco
> models). 
> 4. Tried an OpenVox A400P PCI card and a Sangoma U100 USB
> adapter as analogue-to-digital interfaces.

Have you run fxotune?  I remember doing that when we had analog
lines.  You'd have to look up how--maybe just in the fxotune man page.


-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0


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

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

Re: [asterisk-users] asterisk and meetme

[Matt Hamilton]

> ConfBridge is the preferred conference application in Asterisk 10+.  While
> MeetMe is currently deprecated, you can still enable it and run it in
> Asterisk 10+.

What's going to happen to SLA (which is heavily integrated with MeetMe)? Will 
the functionality be ported to ConfBridge?

Thanks,
Matt




> Date: Fri, 10 Aug 2012 08:34:32 -0500
> From: mjor...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] asterisk and meetme
> 
> 
> 
> - Original Message -
> > From: "Jerry Geis" 
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> > 
> > Sent: Friday, August 10, 2012 8:25:54 AM
> > Subject: Re: [asterisk-users] asterisk and meetme
> > 
> > On 08/10/2012 09:00 AM, Jerry Geis wrote:
> 
> > My bad - "make menuconfig" was not coming up as my window was too
> > small,
> > was confused as the help page I was on for asterisk 10 and meetme did
> > not say
> > anything about it being deprecated (as menuconfig does).
> 
> Which help page was it?  If its the application description on the Asterisk
> wiki, you're right - those don't currently display the 'support' status of
> the module that the application resides in.
> 
> You can find out all of the support statuses of the various modules here:
> 
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
>  
> > So meetme is deprecated in asterisk 10. Looks like I need to move to
> > app_confbridge.
> 
> ConfBridge is the preferred conference application in Asterisk 10+.  While
> MeetMe is currently deprecated, you can still enable it and run it in
> Asterisk 10+.
>  
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
  --
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] iCall service any good?

[Bruce B]

Hi everyone,

We are getting cotinueous error messages over the past few days from iCall:

  -- Called iCall/01144
-- Got SIP response 500 "Server internal failure" back from
72.249.14.242

Is this something everyone else is getting? They are very bad at support
and I am not sure if it's their servers or my Asterisk server that is
causing the issue.

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

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

Re: [asterisk-users] Question on app_confbridge

[Jerry Geis]

On 08/10/2012 11:23 AM, Jerry Geis wrote:

I have a profile in confbridge
[MessageNetConfBridge]
and more...

Asterisk is reading it at startup.
[1;30m  == ^[[0mParsing '/etc/asterisk/confbridge.conf': ^[[1;30m  == 
^[[0mFound
^[[1;30m ^[[0mapp_confbridge.so => (^[[0;33mConference Bridge 
Application^[[0m)


When I try to use it I get a warning about
 WARNING[20678] app_confbridge.c: Conference bridge profile  
MessageNetConfBridge does not exist



My dialplan looks like:
exten => new_app_confbridge,1,ConfBridge(${agi_pa_meetme}, 
MessageNetConfBridge)


Not sure how its not picking it up?

What did I miss?

Jerry

Clearly I have conf bridge:
confbridge show profile bridge MessageNetConfBridge

Name: MessageNetConfBridge
Internal Sample Rate: 8000
Mixing Interval:  Default 20ms
Record Conference:no
Record File:  Auto Generated
Max Members:  No Limit
Video Mode:   no video
sound_join:   confbridge-join
sound_leave:  confbridge-leave
sound_only_person:conf-onlyperson
sound_has_joined: conf-hasjoin
sound_has_left:   conf-hasleft
sound_kicked: conf-kicked
sound_muted:  conf-muted
sound_unmuted:conf-unmuted
sound_there_are:  conf-thereare
sound_other_in_party: conf-otherinparty
sound_place_into_conference: conf-placeintoconf
sound_wait_for_leader:   conf-waitforleader
sound_leader_has_left:   conf-leaderhasleft
sound_get_pin:conf-getpin
sound_invalid_pin:conf-invalidpin
sound_locked: conf-locked
sound_unlocked_now:   conf-unlockednow
sound_lockednow:  conf-lockednow
sound_error_menu: conf-errormenu

So why is it telling me I do not?

jerry

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

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

Re: [asterisk-users] Multi-tenant IVR

[Carlos Alvarez]

On Fri, Aug 10, 2012 at 2:49 AM, Kannan  wrote:

> In contrast, hosted IVR will have only one number dedicated to a business,
> and the business can maintain the call flow and sound files. The system
> will integrate with their CRM and offer personalized services to the
> customers of the business. And, of course, the system will have the support
> to connect to the PBX of the business, should the customer of the business
> selects to talk to the customer care agent of the business. That is our
> system won’t be used for the communication between the extensions of the
> business.
>

In order to do CRM or other client-side application integration, you'll
need to create your own connectivity into Asterisk.  The security in
Asterisk's remote interfaces isn't great, and I'd say you need to develop
some middleware that handles security and also makes it more robust.
Letting the customers manage their changes would also require some
interface you develop, and that part can get very complex because of things
like dialplan reloading.  We do not allow client access to our hosted
PBX/IVR systems, so I can't advise you on that.


-- 
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Question on app_confbridge

[Jerry Geis]

I have a profile in confbridge
[MessageNetConfBridge]
and more...

Asterisk is reading it at startup.
[1;30m  == ^[[0mParsing '/etc/asterisk/confbridge.conf': ^[[1;30m  == 
^[[0mFound
^[[1;30m ^[[0mapp_confbridge.so => (^[[0;33mConference Bridge 
Application^[[0m)


When I try to use it I get a warning about
 WARNING[20678] app_confbridge.c: Conference bridge profile  
MessageNetConfBridge does not exist



My dialplan looks like:
exten => new_app_confbridge,1,ConfBridge(${agi_pa_meetme}, 
MessageNetConfBridge)


Not sure how its not picking it up?

What did I miss?

Jerry

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

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

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

[Sazzad]

>
> But still contact your ISP and get them to un-block your port 5060.  You
> paid
> them for an Internet connection; and an Internet connection means *all*
> ports,
> not just *some* ports.
>

There has been lots misuses of VOIP here and government policies are
strange. Instead, lots of illegal VOIP businesses run. As I'm not a
POWERFUL guy, I can't simply ask my ISP to unblock it.

Law and constitution, were a means to protect us. (Now, don't smile. I know
it amuses you too.)

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

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

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

[Sazzad]

>
> Oh, I see - check if your country blocks the SIP port 5060 ? try changing
> the default poert from 5060 to something else like  and then try this.
> I think your ISP is blocking the SIP.
>
> HEY!!

You rock. Yeah it seems this is the main problem. I tried with  and saw
the message at instance. Then I changed to ensure to port 5060 and it never
reaches.Strange.
I'll make necessary changes to reconfigure in Asterisk, and let you know if
I can register my phone. Thank you very much.

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

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

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

[Sazzad]

>
> 1.  Can you send UDP packets across your LAN?
> If not, check your client machine.
>
> Ok, I can't register a SIP phone from another PC to my Asterisk server
under same net. Can you tell me one thing? When I call to 127.0.0.1 to my
local Asterisk server it works. When I call to the IP of my ethernet card,
it doesn't work. Should it ought to work?

2.  Can you send UDP packets across the Internet to another host?  (This may
> require the co-operation of another party.)
>
> No, I can't. I tried to send UDP packet to my Rackspace server from my PC
using python code. And also using nc. So far I remember the default TCP
packets worked, but when I tried to send UDP using -u, it didn't work.

> If you can send UDP across your LAN and across the Internet but not to your
> own box in Rackspace, it must be a routing issue at the Rackspace end.
>
> I hope it was true. But isn't it too unlikely?

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

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

Re: [asterisk-users] asterisk and meetme

[Matthew Jordan]

- Original Message -
> From: "Jerry Geis" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, August 10, 2012 8:25:54 AM
> Subject: Re: [asterisk-users] asterisk and meetme
> 
> On 08/10/2012 09:00 AM, Jerry Geis wrote:

> My bad - "make menuconfig" was not coming up as my window was too
> small,
> was confused as the help page I was on for asterisk 10 and meetme did
> not say
> anything about it being deprecated (as menuconfig does).

Which help page was it?  If its the application description on the Asterisk
wiki, you're right - those don't currently display the 'support' status of
the module that the application resides in.

You can find out all of the support statuses of the various modules here:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
 
> So meetme is deprecated in asterisk 10. Looks like I need to move to
> app_confbridge.

ConfBridge is the preferred conference application in Asterisk 10+.  While
MeetMe is currently deprecated, you can still enable it and run it in
Asterisk 10+.
 
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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

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

Re: [asterisk-users] asterisk and meetme

[Jerry Geis]

On 08/10/2012 09:00 AM, Jerry Geis wrote:

I just downloaded and compiled from source asterisk 10.7.0
after installing and running I tried to do a meetme, did not work.
I looked in the apps/app_meetme* and there is only the C file, there 
is no .o

seems like it did not compile.

Is that a new default behavior?

Looking for the trick to get it compiled?

DAHDI is installed my system and works fine.

I dont see anything meetme related in configure.

What did I miss?

Jerry

My bad - "make menuconfig" was not coming up as my window was too small,
was confused as the help page I was on for asterisk 10 and meetme did 
not say

anything about it being deprecated (as menuconfig does).

So meetme is deprecated in asterisk 10. Looks like I need to move to 
app_confbridge.


Jerry

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

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

Re: [asterisk-users] Multi-tenant IVR

[Mitul Limbani]

What you want can be done by OpenVBX, why dont you try exploring that model
?

Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422




On Fri, Aug 10, 2012 at 3:19 PM, Kannan  wrote:

> Hi Carlos,
>
> The idea is this. We are planning to offer customized version of Asterisk
> for specialized purposes. When we offer hosted PBX, using multi-tenancy
> support, it is just going to be PBX, as opposed to a fully blown IVR. It
> will have automated attendant feature, but not IVR.
>
>
>
> In contrast, hosted IVR will have only one number dedicated to a business,
> and the business can maintain the call flow and sound files. The system
> will integrate with their CRM and offer personalized services to the
> customers of the business. And, of course, the system will have the support
> to connect to the PBX of the business, should the customer of the business
> selects to talk to the customer care agent of the business. That is our
> system won’t be used for the communication between the extensions of the
> business.
>
>
> Do you have any reservations on this?
>
>
> Regards,
>
> Kannan.
>
>
>
>
>  On Thu, Aug 9, 2012 at 11:38 PM, Carlos Alvarez wrote:
>
>>
>>
>> On Thu, Aug 9, 2012 at 10:59 AM, Kannan  wrote:
>>
>>> Is should be possible to CONFGURE Asterisk as a multi-tenant IVR server
>>> right? Like tweaking configuration to configure a multi-tenant PBX with
>>> Asterisk.
>>>
>>
>> I don't know why you make a distinction between a multi-tenant IVR and a
>> multi-tenant PBX.  The IVR would just be in tenant contexts just like all
>> other features.
>>
>>
>> --
>> Carlos Alvarez
>> TelEvolve
>> 602-889-3003
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] asterisk and meetme

[Jerry Geis]

I just downloaded and compiled from source asterisk 10.7.0
after installing and running I tried to do a meetme, did not work.
I looked in the apps/app_meetme* and there is only the C file, there is 
no .o

seems like it did not compile.

Is that a new default behavior?

Looking for the trick to get it compiled?

DAHDI is installed my system and works fine.

I dont see anything meetme related in configure.

What did I miss?

Jerry

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

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

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

[A J Stiles]

On Friday 10 August 2012, Patrick Lists wrote:
> On 10-08-12 10:12, SamyGo wrote:
> > Oh, I see - check if your country blocks the SIP port 5060 ? try
> > changing the default poert from 5060 to something else like  and
> > then try this.
> > I think your ISP is blocking the SIP.
> 
> If that is the case, setup an IAX connection and see if that works.

But still contact your ISP and get them to un-block your port 5060.  You paid 
them for an Internet connection; and an Internet connection means *all* ports, 
not just *some* ports.

-- 
AJS

Answers come *after* questions.

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

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

[asterisk-users] chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway!

[equis software]

Hi all.
I have this problem with my Digium 2E1 card and PRI, for hours It works
well, with some meesages...

[Aug 10 09:20:31] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1

But PRI continue uphours later... PRI go down.
I thought the problem was in the telco, but the strange thing is that I
have a loop cable in the second E1 and when I scan both E1 are  with
alarms=LMFA/OK and i have only the first E1 connected to the telco!!

messages
[Aug 10 09:20:31] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:32] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6)
on Primary D-channel of span 1
[Aug 10 09:20:36] WARNING[32270] chan_dahdi.c: No D-channels available!
Using Primary channel 16 as D-channel anyway!

gentoo1 ~ # dahdi_scan
[1]
active=yes
alarms=LMFA/OK
description=T2XXP (PCI) Card 0 Span 1
name=TE2/0/1
manufacturer=Digium
devicetype=Wildcard TE220 (5th Gen)
location=Board ID Switch 0
basechan=1
totchans=31
irq=16
type=digital-E1
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI,HDB3
framing_opts=CCS,CRC4
coding=HDB3
framing=CCS/CRC4
[2]
active=yes
alarms=LMFA/OK
description=T2XXP (PCI) Card 0 Span 2
name=TE2/0/2
manufacturer=Digium
devicetype=Wildcard TE220 (5th Gen)
location=Board ID Switch 0
basechan=32
totchans=31
irq=16
type=digital-E1
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI,HDB3
framing_opts=CCS,CRC4
coding=HDB3
framing=CCS/CRC4
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

[Patrick Lists]

On 10-08-12 10:12, SamyGo wrote:

Oh, I see - check if your country blocks the SIP port 5060 ? try
changing the default poert from 5060 to something else like  and
then try this.
I think your ISP is blocking the SIP.


If that is the case, setup an IAX connection and see if that works.

Regards,
Patrick



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

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

Re: [asterisk-users] Multi-tenant IVR

[Kannan]

Hi Carlos,

The idea is this. We are planning to offer customized version of Asterisk
for specialized purposes. When we offer hosted PBX, using multi-tenancy
support, it is just going to be PBX, as opposed to a fully blown IVR. It
will have automated attendant feature, but not IVR.



In contrast, hosted IVR will have only one number dedicated to a business,
and the business can maintain the call flow and sound files. The system
will integrate with their CRM and offer personalized services to the
customers of the business. And, of course, the system will have the support
to connect to the PBX of the business, should the customer of the business
selects to talk to the customer care agent of the business. That is our
system won’t be used for the communication between the extensions of the
business.


Do you have any reservations on this?


Regards,

Kannan.




On Thu, Aug 9, 2012 at 11:38 PM, Carlos Alvarez wrote:

>
>
> On Thu, Aug 9, 2012 at 10:59 AM, Kannan  wrote:
>
>> Is should be possible to CONFGURE Asterisk as a multi-tenant IVR server
>> right? Like tweaking configuration to configure a multi-tenant PBX with
>> Asterisk.
>>
>
> I don't know why you make a distinction between a multi-tenant IVR and a
> multi-tenant PBX.  The IVR would just be in tenant contexts just like all
> other features.
>
>
> --
> Carlos Alvarez
> TelEvolve
> 602-889-3003
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

[SamyGo]

Oh, I see - check if your country blocks the SIP port 5060 ? try changing
the default poert from 5060 to something else like  and then try this.
I think your ISP is blocking the SIP.

On Fri, Aug 10, 2012 at 1:10 PM, A J Stiles
wrote:

> On Thursday 09 August 2012, Sazzad wrote:
> > Hi,
> >
> > I've successfully setup Asterisk on my local PC and can make call using
> > Twinkle to the server. But, I cannot call to my Asterisk server at
> > Rackspace. .
> >
> > My question is how can I troubleshoot this scenario? (Is this question
> > within the scope of this mailing list?)
>
> 1.  Can you send UDP packets across your LAN?
> If not, check your client machine.
>
> 2.  Can you send UDP packets across the Internet to another host?  (This
> may
> require the co-operation of another party.)
> If not, check your router.
>
> If you can send UDP across your LAN and across the Internet but not to your
> own box in Rackspace, it must be a routing issue at the Rackspace end.
>
>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

[A J Stiles]

On Thursday 09 August 2012, Sazzad wrote:
> Hi,
> 
> I've successfully setup Asterisk on my local PC and can make call using
> Twinkle to the server. But, I cannot call to my Asterisk server at
> Rackspace. .
> 
> My question is how can I troubleshoot this scenario? (Is this question
> within the scope of this mailing list?)

1.  Can you send UDP packets across your LAN?
If not, check your client machine.

2.  Can you send UDP packets across the Internet to another host?  (This may 
require the co-operation of another party.)
If not, check your router.

If you can send UDP across your LAN and across the Internet but not to your 
own box in Rackspace, it must be a routing issue at the Rackspace end.


-- 
AJS

Answers come *after* questions.

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

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

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

[Sazzad]

>
> 1-a:Are your SIP packets from PC/SoftPhone reaching the server !! On
> Asterisk CLI execute "*CLI>sip set debug on"
>
> Yeah I've done that, and no UDP packets are reaching my Asterisk server
and neither I can catch any UDP packets at my server using nc -u.

 - are you even able to ping
> your server.
>
> Yes.

1-b: check if you've iptables ON on your server? "iptables -L" if its ON
> then you just flush it "iptables -F" and then see packets reaching your
> server !
>
> There is current no rule as I can see from the output. Why should I need
to configure iptables, anyway? I've read
this,
it mentions iptables to handle something related to RTP packets. But I
can't even get any UDP packets.

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

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