Re: [asterisk-users] Multiple Asterisk

2011-04-29 Thread RAJNIKANT VANZA
Hi,

If u want to setup for 4500 or more phone then better to user OpenSER +
Asterisk.

OpenSER easily work for 10,000 calls.

You need to setup one server for OpenSER and all phone register on this
server. You need to write routing logic in OpenSER server to call connect
and if u need to play media then forward to destination asterisk server.

1 OpenSER server + Asterisk server for each location.


-- 
Best Regards,

Rajnikant Vanza
Software Engineer
---
Working On Linux,C/C++,VoIP,Asterisk Technology


On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes sf.ri...@gmail.comwrote:

 No one ?

 Other thing, i was reading about asterisk realtime, it can be configured to
 have multiple asterisk conectted to the same database? But how would it know
 in wich host are the number??

 Thanks!

 []'sf.rique


 On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes sf.ri...@gmail.comwrote:

 I am reading about, and some people are saying that openser is better for
 biger envoriments, and dundi is fine for smal envoriments, does anyone have
 any info about it ?

 We have now about 4500 convencional phones and we gonna expand a lot.

 So,

 OpenSER vs DUNDi ?

 I guess i will use Asterisk RealTime also right ?


 []'sf.rique


 On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham rizwanhas...@gmail.comwrote:

 Here is a better link for DUNDi


 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/

 skip the part which you know already


 On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes 
 sf.ri...@gmail.comwrote:


 []'sf.rique


 On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger 
 pabelan...@digium.comwrote:

 On 11-03-15 06:19 PM, Henrique Fernandes wrote:

 Have many diferenet locations that have convencional phones that need
 to
 call others locations with convencional phones. And we can not change
 this,
 I was reading and asterisk cannot handle it self this kind of setup,
 it
 needs an separated serrver to control and routers the calls to this
 poins
 right ?

 So can you guys give any help ? I guess asterisk with SER could do the
 job ?

  I don't believe SER will help you in the setup (see below).


  So my question is how do i make the 2 PABX with asterisk talk to  each
 other?  Do i need only 2 asterisk with digium or i need one server
 with SER
 to maki it happen ? There is another program that does what i am
 looking for
 ?

  If you require local hardware for each site, then you can install
 Asterisk at each location.  You can then interconnect them using IAX2 or
 SIP, additionally you can use DUNDi in your dialplans to share information
 before the Asterisk boxes.


 Thanks!

 I had heard some thing about DUNDi but now i am reading i guess it is
 what i need!

 I am guessing i can use both IAX2 and SIP i read something about H.323

 So i am gonna see which one is best to conect the Asterisk PBX if i am
 not able to use bot SIP and IAX2

 Thanks!

 here is a link that explains better what DUNDi is!

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


 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org

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 --
 Best Ragards
 Rizwan Qureshi
 VoIP/Asterisk Engineer
 Axvoice Inc.

 V: +92 (0)  6767 26
 E: rizwanhas...@gmail.com
 W: www.axvoice.com


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-29 Thread Olle E. Johansson
 
 
 I don't think there's anything inherently wrong with the bug tracking system. 
  It's more of a resource issue with many conflicting priorities.  Officially 
 letting off some of the pressure from older branches does help.  I would like 
 to be making faster progress through bug reports and patches.  I do have an 
 open position for another full time Asterisk developer at Digium in case 
 anyone is interested.  :-)
 

I agree with Russell here, we have resource issues in the bug tracker but 
that's nothing that can be solved by another piece of software. If you have 
issues that is not handled timely, why don't you spend some time with other 
issues to help out? Surely there are issues where you can give a helping hand.

In answer to an earlier email that I felt was kind of attacking me I would like 
to point out that I am very happy and grateful about the resources that Digium 
put in the project, and continue to do. Just to clarify that this discussion 
was not about trying to paint Digium as a company as evil - which I was accused 
of. Digium is a very old business partner to my company and we've done great 
together. That doesn't mean we can't critizise each other or not want to 
discuss issues in the open.

To answer another attack, I have been contributing code and bug fixes to both 
1.8 and trunk. Most of my code exist in versions for trunk and 1.4. Customers 
pay me for 1.4, I forward port it to trunk when I have time and resources over. 
It's not a personal choice that most of my development work still is based on 
1.4. Of course I would love being doing development freely, creating great new 
code for the new release. There's a lot of stuff to do in Asterisk trunk, but 
no one out there that wants to put resources towards it in my direction. 
Asterisk trunk development is sadly too far away from my customers current 
business. The 1.6.x release schedule widened that gap and we need to discuss 
how to close the gap again. We do not need a large number of maintained 
releases between the long term support releases.

So far I haven't seen more than a few people that chimes in to this discussion 
saying we need to have 1.4 open, I haven't seen many people running 1.8 in 
production either. I have seen a lot of important issues being reported with 
1.8 which to me confirms that it's still not ready.

I have been working in commercial software companies for a long period in my 
life. A product manager that called for end-of-life of the 1.4 release at this 
stage would be out of a job very soon. Migrating a customer base from one 
version to another is very, very hard. It seems much harder in telecom software 
than in the rest of the software world. We need to continue to work on 1.8 and 
do a lot of marketing for upgrading as soon as we're comfortable with it and 
have resources to manage the bug reports that will come in. We really need to 
push and shove. What I can't do to my customers is forcing them to upgrade to 
something that doesn't work. Customer will simply stop paying me if I do.

I will not continue to push this issue, just realize that I will have to manage 
my own 1.4 branch fixing the issues that affect my customers, which will 
exclude management of a lot of modules that are not used at all in our 
installations. As I said before, I have no resources to support all of the code 
base for everyone. That's just life, painful as it is. In the ideal world, 
there would be resources to help everyone. Unfortunately, I still have to have 
money to bring home at the end of the day.

Thanks for a very good discussion. As usual, I learned a lot from it. Keep 
reporting issues so that all of us can move forward to new releases.

Feel free to contact me off-list if you want to discuss this further.

/O


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-29 Thread Olle E. Johansson

29 apr 2011 kl. 01.49 skrev Leif Madsen:

 Well the issue is that we currently have over 900 open issues in the Asterisk
 project alone, and with only one primary bug marshal (myself) sometimes things
 accidentally get closed if it looks like a configuration issue.

What's the reason that we only have one bug marshal? We used to ask people to 
become bug marshals to help,
but the last I heard you and Russell did not want community marshals. What went 
wrong with that? Wasn't it any help?

/O
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-29 Thread Administrator TOOTAI

Le 29/04/2011 00:42, Russell Bryant a écrit :

- Original Message -

Sure. Please follow the 2 next stories:

- had a customer running 1.4.26 We upgraded to a new server and
installed 1.4.39, last version at this time. Bang: voicemail doesn't
work as it should, had to fallback to 1.4.26 Customer is still running
this version.
- have 1.4.41 and 1.6.16 which are no more able to use auth keys in
iax
since we update one server from 1.4 to 1.6

Now imagine that 1.4 stays at only security level. For first case we
have 2 options: upgrading for security reasons to last version but
then no more voicemail, or staying with 1.4.26. In the second case,
upgrading both servers to test with 1.8. If it's still not working, it was time
loose beside other problems.

If there are obvious regressions in major functionality such as voicemail, I'm more than 
happy to still consider making fixes for those problems during the security 
maintenance period.  It has to be pretty clear, though, and in this particular 
case, it is.

Can you point to the bug number please?  I want to make sure this voicemail 
problem is resolved as soon as possible.


https://issues.asterisk.org/view.php?id=18998 for the voicemail
https://issues.asterisk.org/view.php?id=18539 for the iax2 auth rsa

--
Daniel

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Re: [asterisk-users] Discussion: 1.8 quality issues

2011-04-29 Thread Olle E. Johansson

28 apr 2011 kl. 16.53 skrev Russell Bryant:

 
 - Original Message -
 PS. Please don't start a discussion about 1.8 quality in this thread,
 that's a separate issue. I just want to know what you think about
 closing 1.4 support now. If you want to discuss 1.8 quality, start a
 new thread. Thanks.
 
 I don't think it's a separate issue at all.  I would like to see discussion 
 of exactly which issues are preventing users from using Asterisk 1.8.  We're 
 trying to shift focus to those issues and get them resolved as quickly and as 
 efficiently as we can so that we can all move forward.
Thanks for ignoring my plea... Please at least change the subject ;-)
 
 Resources are limited.  What is the best use of our time to help ensure the 
 best future?  Where do we want to see the project in the next 6 months to a 
 year?  A primary focus on further solidifying Asterisk 1.8 is what gets us 
 there in my mind.
I agree.
 
 Asterisk 1.4 was released 4.5 years ago.  It mostly just works, and I fully 
 expect many to keep using it until they see a need to migrate.  
If you think it's mostly just works it can't be hard to support it a while 
longer then, can it?

 This process has been likened to when the community moved from Asterisk 1.2 
 to 1.4.  Asterisk 1.8 has been much more stable out of the gate than 1.4, due 
 to many things we have done over the years to increase quality, including:
 
 1) We have adopted peer code reviews as common practice for all non-trivial 
 changes going into Asterisk.  This alone has _greatly_ increased the quality 
 of the code going in.  It is rare that a patch goes up for review where 
 someone doesn't point out some sort of problem.  These problems are found and 
 fixed _much_ faster in the up front review process than if it had been many 
 months later when someone encountered it as a bug in the field.
Agree. But it also puts a significant delay on the process. We have to be very 
careful about that. Having too many branches open in addition to this was a 
pain. With fewer branches I hope it will get better.

 
 2) We have placed an increased emphasis on automated testing efforts.  In 
 addition to building up a lot of test environments inside of Digium, there is 
 now an open source automated testing effort for Asterisk.  There are over 200 
 test cases that run every time anyone touches the code.  This includes 
 complex call scenarios such as transfers and call parking.  These open source 
 test cases touch about 25% of the code (and what it does touch are things we 
 considered some of the most important parts).  That is a huge step forward 
 from where we started.  We are continuing to place more and more resources on 
 this effort to move it forward.
Agree. It's great and we need to continue working on it, because it obviously 
hasn't caught everything we should have caught. I fully agree that it is a 
wonderful system and I've said that many, many times.
 
 Despite comments in this thread, there _are_ many people using Asterisk 1.8 
 in production, including large installations.  The ones with systems working 
 perfectly fine don't tend to make as much noise.  :-)  For those still 
 getting hit by problems, I hope that you can make the time to report them so 
 that we can work with you to get them resolved.
I don't disagree there either. I have only stated that it fails in my and my 
customer's installations. Everyone is using Asterisk in different ways. If it 
did not work anywhere I would be very disappointed.
 
 I started my work on Asterisk as a volunteer 7 years ago and even though it 
 is now my full time job, I still put many personal hours into the project.  I 
 care very deeply about the success of Asterisk.  I truly believe that the 
 steps we have taken with release management are in the best interest of the 
 project.
I understand that you do, I don't think you do things you don't believe in. But 
you do need feedback from production sites to make the best decisions. 

What you bring up here is important but in my world have no relation to the 
decision about 1.4. I understand you want to use development resources in a 
good way, but there are also marketing/business perspectives to consider here. 
I personally don't think closing 1.4 support today is in the best interest of 
the project from a marketing point of view, as I don't believe we have a 
working alternative to offer. I understand we have different opinions about it.

Regards,
/Olle
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[asterisk-users] Hardware Server Configuration/8 or 4 port PRI Card

2011-04-29 Thread Kaushal Shriyan
Hi,

Can someone please recommend me the Hardware Server Configuration/8 or 4
port PRI Card to make Outbound Call at the rate of around 320 outbound
Calls/min ?

Thanks and Regards,

Kaushal
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Re: [asterisk-users] Multiple Asterisk

2011-04-29 Thread Henrique Fernandes
Thnaks a Lot.

So i will look for openser integration with asterisk!

[]'sf.rique


On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA rajniva...@gmail.comwrote:

 Hi,

 If u want to setup for 4500 or more phone then better to user OpenSER +
 Asterisk.

 OpenSER easily work for 10,000 calls.

 You need to setup one server for OpenSER and all phone register on this
 server. You need to write routing logic in OpenSER server to call connect
 and if u need to play media then forward to destination asterisk server.

 1 OpenSER server + Asterisk server for each location.


 --
 Best Regards,

 Rajnikant Vanza
 Software Engineer
 ---
 Working On Linux,C/C++,VoIP,Asterisk Technology


 On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes 
 sf.ri...@gmail.comwrote:

 No one ?

 Other thing, i was reading about asterisk realtime, it can be configured
 to have multiple asterisk conectted to the same database? But how would it
 know in wich host are the number??

 Thanks!

 []'sf.rique


 On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes 
 sf.ri...@gmail.comwrote:

 I am reading about, and some people are saying that openser is better for
 biger envoriments, and dundi is fine for smal envoriments, does anyone have
 any info about it ?

 We have now about 4500 convencional phones and we gonna expand a lot.

 So,

 OpenSER vs DUNDi ?

 I guess i will use Asterisk RealTime also right ?


 []'sf.rique


 On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham 
 rizwanhas...@gmail.comwrote:

 Here is a better link for DUNDi


 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/

 skip the part which you know already


 On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.com
  wrote:


 []'sf.rique


 On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger 
 pabelan...@digium.comwrote:

 On 11-03-15 06:19 PM, Henrique Fernandes wrote:

 Have many diferenet locations that have convencional phones that need
 to
 call others locations with convencional phones. And we can not change
 this,
 I was reading and asterisk cannot handle it self this kind of setup,
 it
 needs an separated serrver to control and routers the calls to this
 poins
 right ?

 So can you guys give any help ? I guess asterisk with SER could do
 the job ?

  I don't believe SER will help you in the setup (see below).


  So my question is how do i make the 2 PABX with asterisk talk to
  each
 other?  Do i need only 2 asterisk with digium or i need one server
 with SER
 to maki it happen ? There is another program that does what i am
 looking for
 ?

  If you require local hardware for each site, then you can install
 Asterisk at each location.  You can then interconnect them using IAX2 or
 SIP, additionally you can use DUNDi in your dialplans to share 
 information
 before the Asterisk boxes.


 Thanks!

 I had heard some thing about DUNDi but now i am reading i guess it is
 what i need!

 I am guessing i can use both IAX2 and SIP i read something about H.323

 So i am gonna see which one is best to conect the Asterisk PBX if i am
 not able to use bot SIP and IAX2

 Thanks!

 here is a link that explains better what DUNDi is!

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


 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org

 --
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 --
 Best Ragards
 Rizwan Qureshi
 VoIP/Asterisk Engineer
 Axvoice Inc.

 V: +92 (0)  6767 26
 E: rizwanhas...@gmail.com
 W: www.axvoice.com


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Re: [asterisk-users] odbc error - server is gone

2011-04-29 Thread Rizwan Hisham
Yes I have it there, here the content of the file:

i think the code is buggy,

here is a comment from the function which generated the error
(ast_odbc_smart_execute in res_odbc.c line 155 )

/* This is a really bad method of trying to correct a dead connection.  It
 * only ever really worked with MySQL.  It will not work with any other
 * database, since most databases prepare their statements on the server,
 * and if you disconnect, you invalidate the statement handle.  Hence, if
 * you disconnect, you're going to fail anyway, whether you try to execute
 * a second time or not.
 */

This function is used all over.

On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan 
sherwood.mcgo...@gmail.com wrote:

 On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.comwrote:

 Hi list,
 yesterday I converted my voicemail.conf to realtime voicemail and also
 configured to store the voicemessages in a database using odbc as described
 here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail and
 here http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
 .
 I am using asterisk 1.4.2 with mysql. I also installed the proper odbc
 driver for mysql on the server. I successfully completed the conversion of a
 lot of voicemail users into db yesterday. But today on the CLI thsi error
 was showing;

 [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute:
 SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
 Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
 [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute:
 SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51
 Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
 [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL
 Execute error!
 [SELECT COUNT(*) FROM voicemessages WHERE dir =
 '/var/spool/asterisk/voicemail/default/1757XXX/INBOX']

 I know that the error is caused due to stale odbc connection with mysql.
 But i want to find out if there is a cure for it. Why the connection went
 stale in the first place also.

 Any ideas?

 --
 Best Ragards
 Rizwan Qureshi
 VoIP/Asterisk Engineer
 Axvoice Inc.

 V: +92 (0)  6767 26
 E: rizwanhas...@gmail.com
 W: www.axvoice.com


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 do you have sanitysql = select 1 configured in res_odbc.ini?

 --
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 Telecommunications and VOIP Consultant


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-- 
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.

V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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Re: [asterisk-users] Multiple Asterisk

2011-04-29 Thread Satish Patel
True, we had setup before openser with asterisk and it works great. I  
have wrote small document on voip-info related my project.


--
Sent from my iPhone

On Apr 29, 2011, at 8:23 AM, Henrique Fernandes sf.ri...@gmail.com  
wrote:



Thnaks a Lot.

So i will look for openser integration with asterisk!

[]'sf.rique


On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA  
rajniva...@gmail.com wrote:

Hi,

If u want to setup for 4500 or more phone then better to user  
OpenSER + Asterisk.


OpenSER easily work for 10,000 calls.

You need to setup one server for OpenSER and all phone register on  
this server. You need to write routing logic in OpenSER server to  
call connect and if u need to play media then forward to destination  
asterisk server.


1 OpenSER server + Asterisk server for each location.


--
Best Regards,

Rajnikant Vanza
Software Engineer
---
Working On Linux,C/C++,VoIP,Asterisk Technology


On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes sf.ri...@gmail.com 
 wrote:

No one ?

Other thing, i was reading about asterisk realtime, it can be  
configured to have multiple asterisk conectted to the same database?  
But how would it know in wich host are the number??


Thanks!

[]'sf.rique


On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes sf.ri...@gmail.com 
 wrote:
I am reading about, and some people are saying that openser is  
better for biger envoriments, and dundi is fine for smal  
envoriments, does anyone have any info about it ?


We have now about 4500 convencional phones and we gonna expand a lot.

So,

OpenSER vs DUNDi ?

I guess i will use Asterisk RealTime also right ?


[]'sf.rique


On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham  
rizwanhas...@gmail.com wrote:

Here is a better link for DUNDi

http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/

skip the part which you know already


On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.com 
 wrote:


[]'sf.rique


On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger  
pabelan...@digium.com wrote:

On 11-03-15 06:19 PM, Henrique Fernandes wrote:
Have many diferenet locations that have convencional phones that  
need to
call others locations with convencional phones. And we can not  
change this,
I was reading and asterisk cannot handle it self this kind of setup,  
it
needs an separated serrver to control and routers the calls to this  
poins

right ?

So can you guys give any help ? I guess asterisk with SER could do  
the job ?


I don't believe SER will help you in the setup (see below).


So my question is how do i make the 2 PABX with asterisk talk to  each
other?  Do i need only 2 asterisk with digium or i need one server  
with SER
to maki it happen ? There is another program that does what i am  
looking for

?

If you require local hardware for each site, then you can install  
Asterisk at each location.  You can then interconnect them using  
IAX2 or SIP, additionally you can use DUNDi in your dialplans to  
share information before the Asterisk boxes.


Thanks!

I had heard some thing about DUNDi but now i am reading i guess it  
is what i need!


I am guessing i can use both IAX2 and SIP i read something about H.323

So i am gonna see which one is best to conect the Asterisk PBX if i  
am not able to use bot SIP and IAX2


Thanks!

here is a link that explains better what DUNDi is!

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


--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
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

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


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_
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--
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.

V: +92 (0)  6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com


--
_
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_

Re: [asterisk-users] Multiple Asterisk

2011-04-29 Thread Henrique Fernandes
Can you post later t he link for it ?

I read alot that page.

[]'sf.rique


On Fri, Apr 29, 2011 at 9:35 AM, Satish Patel satish...@hotmail.com wrote:

 True, we had setup before openser with asterisk and it works great. I have
 wrote small document on voip-info related my project.

 --
 Sent from my iPhone

 On Apr 29, 2011, at 8:23 AM, Henrique Fernandes sf.ri...@gmail.com
 wrote:

 Thnaks a Lot.

 So i will look for openser integration with asterisk!

 []'sf.rique


 On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA  rajniva...@gmail.com
 rajniva...@gmail.com wrote:

 Hi,

 If u want to setup for 4500 or more phone then better to user OpenSER +
 Asterisk.

 OpenSER easily work for 10,000 calls.

 You need to setup one server for OpenSER and all phone register on this
 server. You need to write routing logic in OpenSER server to call connect
 and if u need to play media then forward to destination asterisk server.

 1 OpenSER server + Asterisk server for each location.


 --
 Best Regards,

 Rajnikant Vanza
 Software Engineer
 ---
 Working On Linux,C/C++,VoIP,Asterisk Technology


 On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes sf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 No one ?

 Other thing, i was reading about asterisk realtime, it can be configured
 to have multiple asterisk conectted to the same database? But how would it
 know in wich host are the number??

 Thanks!

 []'sf.rique


 On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes sf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 I am reading about, and some people are saying that openser is better
 for biger envoriments, and dundi is fine for smal envoriments, does anyone
 have any info about it ?

 We have now about 4500 convencional phones and we gonna expand a lot.

 So,

 OpenSER vs DUNDi ?

 I guess i will use Asterisk RealTime also right ?


 []'sf.rique


 On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham rizwanhas...@gmail.com
 rizwanhas...@gmail.com wrote:

 Here is a better link for DUNDi


 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/
 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/

 skip the part which you know already


 On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.com
 sf.ri...@gmail.com wrote:


 []'sf.rique


 On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger pabelan...@digium.com
 pabelan...@digium.com wrote:

 On 11-03-15 06:19 PM, Henrique Fernandes wrote:

 Have many diferenet locations that have convencional phones that
 need to
 call others locations with convencional phones. And we can not
 change this,
 I was reading and asterisk cannot handle it self this kind of setup,
 it
 needs an separated serrver to control and routers the calls to this
 poins
 right ?

 So can you guys give any help ? I guess asterisk with SER could do
 the job ?

  I don't believe SER will help you in the setup (see below).


  So my question is how do i make the 2 PABX with asterisk talk to
  each
 other?  Do i need only 2 asterisk with digium or i need one server
 with SER
 to maki it happen ? There is another program that does what i am
 looking for
 ?

  If you require local hardware for each site, then you can install
 Asterisk at each location.  You can then interconnect them using IAX2 or
 SIP, additionally you can use DUNDi in your dialplans to share 
 information
 before the Asterisk boxes.


 Thanks!

 I had heard some thing about DUNDi but now i am reading i guess it is
 what i need!

 I am guessing i can use both IAX2 and SIP i read something about H.323

 So i am gonna see which one is best to conect the Asterisk PBX if i am
 not able to use bot SIP and IAX2

 Thanks!

 here is a link that explains better what DUNDi is!

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


 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.comhttp://digium.com 
 http://asterisk.orghttp://asterisk.org

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

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



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

 asterisk-users mailing list
 To UNSUBSCRIBE or update options 

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-29 Thread Leif Madsen
On 11-04-29 02:59 AM, Olle E. Johansson wrote:
 
 29 apr 2011 kl. 01.49 skrev Leif Madsen:
 
 Well the issue is that we currently have over 900 open issues in the Asterisk
 project alone, and with only one primary bug marshal (myself) sometimes 
 things
 accidentally get closed if it looks like a configuration issue.
 
 What's the reason that we only have one bug marshal? We used to ask people to 
 become bug marshals to help,
 but the last I heard you and Russell did not want community marshals. What 
 went wrong with that? Wasn't it any help?

Let me clarify, as it was not at all my intention to imply I was the *only* bug
marshal. Poor wording on my part.

There are certainly lots of people that help manage the bug tracker, and I'm
thankful for everyone who responds to issues, asking for the appropriate
information from reporters, and reviewing logs pointing out potential issues
which help developers. It's just I'm the main one handling work flow, making
sure the tracker doesn't get to the point it was when I started working on it
every day (the majority of issues were sitting in 'New' for many weeks).

Sorry if it was implied that I'm the only one working on the bug tracker,
because that is obviously not the case. I am grateful for any help people can
provide, and they are welcome to ask me what they can do to help. I don't
remember a discussion where I would persuade people from not helping :)

I've tried to make the process for moving issues forward as transparent as
possible. Just search Google with site:lists.digium.com leif madsen bug
marshal for a few posts about work flow. Additional information is here:
https://wiki.asterisk.org/wiki/display/AST/Policies+and+Procedures

Information for reporters is here:
https://wiki.asterisk.org/wiki/display/AST/Debugging

Thanks!
Leif.

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Re: [asterisk-users] (no subject)

2011-04-29 Thread Muhammad Usman
you running GSM FWTs with asterisk ?

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

  HI,

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

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

 Thanks

 Abid.

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




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Regards:
(Muhammad υѕмαη )
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Re: [asterisk-users] Multiple Asterisk

2011-04-29 Thread Satish Patel
Don't expect lots of thing because I have just post my basic config  
and method to integrate openser with asterisk and I did that 3 year ago.


http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration.

I would say search on google today lots of material are there and I  
have remembered there is a nice book regarding this. I guess I have  
PDF version of that book I will search and try to find.


--
Sent from my iPhone

On Apr 29, 2011, at 8:40 AM, Henrique Fernandes sf.ri...@gmail.com  
wrote:



Can you post later t he link for it ?

I read alot that page.

[]'sf.rique


On Fri, Apr 29, 2011 at 9:35 AM, Satish Patel  
satish...@hotmail.com wrote:
True, we had setup before openser with asterisk and it works great.  
I have wrote small document on voip-info related my project.


--
Sent from my iPhone

On Apr 29, 2011, at 8:23 AM, Henrique Fernandes sf.ri...@gmail.com  
wrote:



Thnaks a Lot.

So i will look for openser integration with asterisk!

[]'sf.rique


On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA rajniva...@gmail.com 
 wrote:

Hi,

If u want to setup for 4500 or more phone then better to user  
OpenSER + Asterisk.


OpenSER easily work for 10,000 calls.

You need to setup one server for OpenSER and all phone register on  
this server. You need to write routing logic in OpenSER server to  
call connect and if u need to play media then forward to  
destination asterisk server.


1 OpenSER server + Asterisk server for each location.


--
Best Regards,

Rajnikant Vanza
Software Engineer
---
Working On Linux,C/C++,VoIP,Asterisk Technology


On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes sf.ri...@gmail.com 
 wrote:

No one ?

Other thing, i was reading about asterisk realtime, it can be  
configured to have multiple asterisk conectted to the same  
database? But how would it know in wich host are the number??


Thanks!

[]'sf.rique


On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes sf.ri...@gmail.com 
 wrote:
I am reading about, and some people are saying that openser is  
better for biger envoriments, and dundi is fine for smal  
envoriments, does anyone have any info about it ?


We have now about 4500 convencional phones and we gonna expand a lot.

So,

OpenSER vs DUNDi ?

I guess i will use Asterisk RealTime also right ?


[]'sf.rique


On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham rizwanhas...@gmail.com 
 wrote:

Here is a better link for DUNDi

http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/

skip the part which you know already


On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.com 
 wrote:


[]'sf.rique


On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger  
pabelan...@digium.com wrote:

On 11-03-15 06:19 PM, Henrique Fernandes wrote:
Have many diferenet locations that have convencional phones that  
need to
call others locations with convencional phones. And we can not  
change this,
I was reading and asterisk cannot handle it self this kind of  
setup, it
needs an separated serrver to control and routers the calls to this  
poins

right ?

So can you guys give any help ? I guess asterisk with SER could do  
the job ?


I don't believe SER will help you in the setup (see below).


So my question is how do i make the 2 PABX with asterisk talk to   
each
other?  Do i need only 2 asterisk with digium or i need one server  
with SER
to maki it happen ? There is another program that does what i am  
looking for

?

If you require local hardware for each site, then you can install  
Asterisk at each location.  You can then interconnect them using  
IAX2 or SIP, additionally you can use DUNDi in your dialplans to  
share information before the Asterisk boxes.


Thanks!

I had heard some thing about DUNDi but now i am reading i guess it  
is what i need!


I am guessing i can use both IAX2 and SIP i read something about H. 
323


So i am gonna see which one is best to conect the Asterisk PBX if i  
am not able to use bot SIP and IAX2


Thanks!

here is a link that explains better what DUNDi is!

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


--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
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


--
_
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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 

Re: [asterisk-users] Multiple Asterisk

2011-04-29 Thread Henrique Fernandes
Thanks!

Would apreciate the book!

But i am already researching

[]'sf.rique


On Fri, Apr 29, 2011 at 10:10 AM, Satish Patel satish...@hotmail.comwrote:

 Don't expect lots of thing because I have just post my basic config and
 method to integrate openser with asterisk and I did that 3 year ago.

 http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration
 http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration.

 I would say search on google today lots of material are there and I have
 remembered there is a nice book regarding this. I guess I have PDF version
 of that book I will search and try to find.

 --
 Sent from my iPhone

 On Apr 29, 2011, at 8:40 AM, Henrique Fernandes sf.ri...@gmail.com
 wrote:

 Can you post later t he link for it ?

 I read alot that page.

 []'sf.rique


 On Fri, Apr 29, 2011 at 9:35 AM, Satish Patel  satish...@hotmail.com
 satish...@hotmail.com wrote:

 True, we had setup before openser with asterisk and it works great. I have
 wrote small document on voip-info related my project.

 --
 Sent from my iPhone

 On Apr 29, 2011, at 8:23 AM, Henrique Fernandes  sf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 Thnaks a Lot.

 So i will look for openser integration with asterisk!

 []'sf.rique


 On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA  
 rajniva...@gmail.comrajniva...@gmail.com
 rajniva...@gmail.com wrote:

 Hi,

 If u want to setup for 4500 or more phone then better to user OpenSER +
 Asterisk.

 OpenSER easily work for 10,000 calls.

 You need to setup one server for OpenSER and all phone register on this
 server. You need to write routing logic in OpenSER server to call connect
 and if u need to play media then forward to destination asterisk server.

 1 OpenSER server + Asterisk server for each location.


 --
 Best Regards,

 Rajnikant Vanza
 Software Engineer
 ---
 Working On Linux,C/C++,VoIP,Asterisk Technology


 On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes 
 sf.ri...@gmail.comsf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 No one ?

 Other thing, i was reading about asterisk realtime, it can be configured
 to have multiple asterisk conectted to the same database? But how would it
 know in wich host are the number??

 Thanks!

 []'sf.rique


 On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes 
 sf.ri...@gmail.comsf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 I am reading about, and some people are saying that openser is better
 for biger envoriments, and dundi is fine for smal envoriments, does anyone
 have any info about it ?

 We have now about 4500 convencional phones and we gonna expand a lot.

 So,

 OpenSER vs DUNDi ?

 I guess i will use Asterisk RealTime also right ?


 []'sf.rique


 On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham 
 rizwanhas...@gmail.comrizwanhas...@gmail.com
 rizwanhas...@gmail.com wrote:

 Here is a better link for DUNDi


 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/
 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/

 skip the part which you know already


 On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes 
 sf.ri...@gmail.comsf.ri...@gmail.com
 sf.ri...@gmail.com wrote:


 []'sf.rique


 On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger 
 pabelan...@digium.compabelan...@digium.com
 pabelan...@digium.com wrote:

 On 11-03-15 06:19 PM, Henrique Fernandes wrote:

 Have many diferenet locations that have convencional phones that
 need to
 call others locations with convencional phones. And we can not
 change this,
 I was reading and asterisk cannot handle it self this kind of
 setup, it
 needs an separated serrver to control and routers the calls to this
 poins
 right ?

 So can you guys give any help ? I guess asterisk with SER could do
 the job ?

  I don't believe SER will help you in the setup (see below).


  So my question is how do i make the 2 PABX with asterisk talk to
  each
 other?  Do i need only 2 asterisk with digium or i need one server
 with SER
 to maki it happen ? There is another program that does what i am
 looking for
 ?

  If you require local hardware for each site, then you can install
 Asterisk at each location.  You can then interconnect them using IAX2 
 or
 SIP, additionally you can use DUNDi in your dialplans to share 
 information
 before the Asterisk boxes.


 Thanks!

 I had heard some thing about DUNDi but now i am reading i guess it is
 what i need!

 I am guessing i can use both IAX2 and SIP i read something about
 H.323

 So i am gonna see which one is best to conect the Asterisk PBX if i
 am not able to use bot SIP and IAX2

 Thanks!

 here is a link that explains better what DUNDi is!

 http://www.voip-info.org/wiki/view/DUNDihttp://www.voip-info.org/wiki/view/DUNDi
 http://www.voip-info.org/wiki/view/DUNDi


 --
 Paul Belanger
 Digium, Inc. | Software Developer
 

Re: [asterisk-users] Multiple Asterisk

2011-04-29 Thread satish patel

I have sent you book in PM.

-S

Date: Fri, 29 Apr 2011 10:39:56 -0300
From: sf.ri...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multiple Asterisk

Thanks!

Would apreciate the book!

But i am already researching
[]'sf.rique 



On Fri, Apr 29, 2011 at 10:10 AM, Satish Patel satish...@hotmail.com wrote:

Don't expect lots of thing because I have just post my basic config and method 
to integrate openser with asterisk and I did that 3 year ago.


http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration.


I would say search on google today lots of material are there and I have 
remembered there is a nice book regarding this. I guess I have PDF version of 
that book I will search and try to find.


--Sent from my iPhone
On Apr 29, 2011, at 8:40 AM, Henrique Fernandes sf.ri...@gmail.com wrote:


Can you post later t he link for it ?

I read alot that page.

[]'sf.rique 



On Fri, Apr 29, 2011 at 9:35 AM, Satish Patel satish...@hotmail.com wrote:


True, we had setup before openser with asterisk and it works great. I have 
wrote small document on voip-info related my project. 

--Sent from my iPhone


On Apr 29, 2011, at 8:23 AM, Henrique Fernandes sf.ri...@gmail.com wrote:



Thnaks a Lot.

So i will look for openser integration with asterisk!
[]'sf.rique 



On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote:



Hi,

If u want to setup for 4500 or more phone then better to user OpenSER + 
Asterisk.
OpenSER easily work for 10,000 calls.
You need to setup one server for OpenSER and all phone register on this server. 
You need to write routing logic in OpenSER server to call connect and if u need 
to play media then forward to destination asterisk server.




1 OpenSER server + Asterisk server for each location.

-- 
Best Regards,

Rajnikant Vanza
Software Engineer
---




Working On Linux,C/C++,VoIP,Asterisk Technology

On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes sf.ri...@gmail.com wrote:




No one ?

Other thing, i was reading about asterisk realtime, it can be configured to 
have multiple asterisk conectted to the same database? But how would it know in 
wich host are the number??





Thanks!

[]'sf.rique 



On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes sf.ri...@gmail.com wrote:





I am reading about, and some people are saying that openser is better for biger 
envoriments, and dundi is fine for smal envoriments, does anyone have any info 
about it ?

We have now about 4500 convencional phones and we gonna expand a lot.







So, 

OpenSER vs DUNDi ? 

I guess i will use Asterisk RealTime also right ?

[]'sf.rique 



On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham rizwanhas...@gmail.com wrote:







Here is a better link for DUNDi

http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/









skip the part which you know already

On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.com wrote:








[]'sf.rique 



On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger pabelan...@digium.com wrote:









On 11-03-15 06:19 PM, Henrique Fernandes wrote:


Have many diferenet locations that have convencional phones that need to

call others locations with convencional phones. And we can not change this,

I was reading and asterisk cannot handle it self this kind of setup, it

needs an separated serrver to control and routers the calls to this poins

right ?



So can you guys give any help ? I guess asterisk with SER could do the job ?




I don't believe SER will help you in the setup (see below).




So my question is how do i make the 2 PABX with asterisk talk to  each

other?  Do i need only 2 asterisk with digium or i need one server with SER

to maki it happen ? There is another program that does what i am looking for

?




If you require local hardware for each site, then you can install Asterisk at 
each location.  You can then interconnect them using IAX2 or SIP, additionally 
you can use DUNDi in your dialplans to share information before the Asterisk 
boxes.










Thanks!

I had heard some thing about DUNDi but now i am reading i guess it is what i 
need!

I am guessing i can use both IAX2 and SIP i read something about  H.323

So i am gonna see which one is best to conect the Asterisk PBX if i am not able 
to use bot SIP and IAX2










Thanks!

here is a link that explains better what DUNDi is!

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













-- 

Paul Belanger

Digium, Inc. | Software Developer

twitter: pabelanger | IRC: pabelanger (Freenode)

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





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_

-- 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 

Re: [asterisk-users] Multiple Asterisk

2011-04-29 Thread vip killa
could you send me book?

On Fri, Apr 29, 2011 at 9:48 AM, satish patel satish...@hotmail.com wrote:

  I have sent you book in PM.

 -S

 --
 Date: Fri, 29 Apr 2011 10:39:56 -0300
 From: sf.ri...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Multiple Asterisk


 Thanks!

 Would apreciate the book!

 But i am already researching

 []'sf.rique


 On Fri, Apr 29, 2011 at 10:10 AM, Satish Patel satish...@hotmail.comwrote:

 Don't expect lots of thing because I have just post my basic config and
 method to integrate openser with asterisk and I did that 3 year ago.

  http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration
 http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration.

 I would say search on google today lots of material are there and I have
 remembered there is a nice book regarding this. I guess I have PDF version
 of that book I will search and try to find.

 --
 Sent from my iPhone

 On Apr 29, 2011, at 8:40 AM, Henrique Fernandes sf.ri...@gmail.com
 wrote:

 Can you post later t he link for it ?

 I read alot that page.

 []'sf.rique


 On Fri, Apr 29, 2011 at 9:35 AM, Satish Patel  satish...@hotmail.com
 satish...@hotmail.com wrote:

 True, we had setup before openser with asterisk and it works great. I have
 wrote small document on voip-info related my project.

 --
 Sent from my iPhone

 On Apr 29, 2011, at 8:23 AM, Henrique Fernandes  sf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 Thnaks a Lot.

 So i will look for openser integration with asterisk!

 []'sf.rique


 On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA  
 rajniva...@gmail.comrajniva...@gmail.com
 rajniva...@gmail.com wrote:

 Hi,

 If u want to setup for 4500 or more phone then better to user OpenSER +
 Asterisk.

 OpenSER easily work for 10,000 calls.

 You need to setup one server for OpenSER and all phone register on this
 server. You need to write routing logic in OpenSER server to call connect
 and if u need to play media then forward to destination asterisk server.

 1 OpenSER server + Asterisk server for each location.


 --
 Best Regards,

 Rajnikant Vanza
 Software Engineer
 ---
 Working On Linux,C/C++,VoIP,Asterisk Technology


 On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes 
 sf.ri...@gmail.comsf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 No one ?

 Other thing, i was reading about asterisk realtime, it can be configured to
 have multiple asterisk conectted to the same database? But how would it know
 in wich host are the number??

 Thanks!

 []'sf.rique


 On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes  
 sf.ri...@gmail.comsf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 I am reading about, and some people are saying that openser is better for
 biger envoriments, and dundi is fine for smal envoriments, does anyone have
 any info about it ?

 We have now about 4500 convencional phones and we gonna expand a lot.

 So,

 OpenSER vs DUNDi ?

 I guess i will use Asterisk RealTime also right ?


 []'sf.rique


 On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham  
 rizwanhas...@gmail.comrizwanhas...@gmail.com
 rizwanhas...@gmail.com wrote:

 Here is a better link for DUNDi


 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/
 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/

 skip the part which you know already


 On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes  
 sf.ri...@gmail.comsf.ri...@gmail.com
 sf.ri...@gmail.com wrote:


 []'sf.rique


 On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger  
 pabelan...@digium.compabelan...@digium.com
 pabelan...@digium.com wrote:

 On 11-03-15 06:19 PM, Henrique Fernandes wrote:

 Have many diferenet locations that have convencional phones that need to
 call others locations with convencional phones. And we can not change this,
 I was reading and asterisk cannot handle it self this kind of setup, it
 needs an separated serrver to control and routers the calls to this poins
 right ?

 So can you guys give any help ? I guess asterisk with SER could do the job
 ?

  I don't believe SER will help you in the setup (see below).


  So my question is how do i make the 2 PABX with asterisk talk to  each
 other?  Do i need only 2 asterisk with digium or i need one server with SER
 to maki it happen ? There is another program that does what i am looking
 for
 ?

  If you require local hardware for each site, then you can install Asterisk
 at each location.  You can then interconnect them using IAX2 or SIP,
 additionally you can use DUNDi in your dialplans to share information before
 the Asterisk boxes.


 Thanks!

 I had heard some thing about DUNDi but now i am reading i guess it is what
 i need!

 I am guessing i can use both IAX2 and SIP i read something about H.323

 So i am gonna see which one is 

Re: [asterisk-users] Multiple Asterisk

2011-04-29 Thread Henrique Fernandes
Thanks i got it

Another think you may know.

Openser have been forked into opensip and kamailio does you have anyidea
wich one is better ?

I guess i will start with opensips, becasue old openser.org point to there.

Thanks again!

[]'sf.rique


On Fri, Apr 29, 2011 at 10:49 AM, vip killa vipki...@gmail.com wrote:

 could you send me book?


 On Fri, Apr 29, 2011 at 9:48 AM, satish patel satish...@hotmail.comwrote:

  I have sent you book in PM.

 -S

 --
 Date: Fri, 29 Apr 2011 10:39:56 -0300
 From: sf.ri...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Multiple Asterisk


 Thanks!

 Would apreciate the book!

 But i am already researching

 []'sf.rique


 On Fri, Apr 29, 2011 at 10:10 AM, Satish Patel satish...@hotmail.comwrote:

 Don't expect lots of thing because I have just post my basic config and
 method to integrate openser with asterisk and I did that 3 year ago.

  http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration
 http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration.

 I would say search on google today lots of material are there and I have
 remembered there is a nice book regarding this. I guess I have PDF version
 of that book I will search and try to find.

 --
 Sent from my iPhone

 On Apr 29, 2011, at 8:40 AM, Henrique Fernandes sf.ri...@gmail.com
 wrote:

 Can you post later t he link for it ?

 I read alot that page.

 []'sf.rique


 On Fri, Apr 29, 2011 at 9:35 AM, Satish Patel  satish...@hotmail.com
 satish...@hotmail.com wrote:

 True, we had setup before openser with asterisk and it works great. I have
 wrote small document on voip-info related my project.

 --
 Sent from my iPhone

 On Apr 29, 2011, at 8:23 AM, Henrique Fernandes  sf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 Thnaks a Lot.

 So i will look for openser integration with asterisk!

 []'sf.rique


 On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA  
 rajniva...@gmail.comrajniva...@gmail.com
 rajniva...@gmail.com wrote:

 Hi,

 If u want to setup for 4500 or more phone then better to user OpenSER +
 Asterisk.

 OpenSER easily work for 10,000 calls.

 You need to setup one server for OpenSER and all phone register on this
 server. You need to write routing logic in OpenSER server to call connect
 and if u need to play media then forward to destination asterisk server.

 1 OpenSER server + Asterisk server for each location.


 --
 Best Regards,

 Rajnikant Vanza
 Software Engineer
 ---
 Working On Linux,C/C++,VoIP,Asterisk Technology


 On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes 
 sf.ri...@gmail.comsf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 No one ?

 Other thing, i was reading about asterisk realtime, it can be configured
 to have multiple asterisk conectted to the same database? But how would it
 know in wich host are the number??

 Thanks!

 []'sf.rique


 On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes 
 sf.ri...@gmail.comsf.ri...@gmail.com
 sf.ri...@gmail.com wrote:

 I am reading about, and some people are saying that openser is better for
 biger envoriments, and dundi is fine for smal envoriments, does anyone have
 any info about it ?

 We have now about 4500 convencional phones and we gonna expand a lot.

 So,

 OpenSER vs DUNDi ?

 I guess i will use Asterisk RealTime also right ?


 []'sf.rique


 On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham  
 rizwanhas...@gmail.comrizwanhas...@gmail.com
 rizwanhas...@gmail.com wrote:

 Here is a better link for DUNDi


 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/
 http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/

 skip the part which you know already


 On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes 
 sf.ri...@gmail.comsf.ri...@gmail.com
 sf.ri...@gmail.com wrote:


 []'sf.rique


 On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger  
 pabelan...@digium.compabelan...@digium.com
 pabelan...@digium.com wrote:

 On 11-03-15 06:19 PM, Henrique Fernandes wrote:

 Have many diferenet locations that have convencional phones that need to
 call others locations with convencional phones. And we can not change
 this,
 I was reading and asterisk cannot handle it self this kind of setup, it
 needs an separated serrver to control and routers the calls to this poins
 right ?

 So can you guys give any help ? I guess asterisk with SER could do the job
 ?

  I don't believe SER will help you in the setup (see below).


  So my question is how do i make the 2 PABX with asterisk talk to  each
 other?  Do i need only 2 asterisk with digium or i need one server with
 SER
 to maki it happen ? There is another program that does what i am looking
 for
 ?

  If you require local hardware for each site, then you can install
 Asterisk at each location.  You can then interconnect them 

Re: [asterisk-users] Discussion: 1.8 quality issues

2011-04-29 Thread Russell Bryant
- Original Message -
  1) We have adopted peer code reviews as common practice for all
  non-trivial changes going into Asterisk. This alone has _greatly_
  increased the quality of the code going in. It is rare that a patch
  goes up for review where someone doesn't point out some sort of
  problem. These problems are found and fixed _much_ faster in the up
  front review process than if it had been many months later when
  someone encountered it as a bug in the field.

 Agree. But it also puts a significant delay on the process. We have to
 be very careful about that. Having too many branches open in addition
 to this was a pain. With fewer branches I hope it will get better.

Fewer branches should help, but the fact the bar is raised on getting patches 
in due to the peer code review process is no different.  There will always be 
problems with the code developers write.  I view it as if there is a problem in 
the code, it is _much_ less expensive to get it resolved in up front peer 
review as much as possible than later on once users encounter a bug, report it, 
developers debug, fix, and test.  That's the tradeoff.

-- 
Russell Bryant
Digium, Inc.   |   Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806  -  USA
www.digium.com  -=-  www.asterisk.org -=- blogs.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] Multiple Asterisk

2011-04-29 Thread satish patel

I never worked on kamailio but its pretty similar to OpenSER. I would say 
OpenSIP would be good and on internet there are lots of comparison regarding 
this topic. 

One more thing OpenSER is pretty simple because in configuration its using SIP 
messages. If you have good knowledge of SIP protocol then you can easily play 
with config file and achieve your goal 

Best Of luck..

-S 

Date: Fri, 29 Apr 2011 10:55:50 -0300
From: sf.ri...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multiple Asterisk

Thanks i got it

Another think you may know.

Openser have been forked into opensip and kamailio does you have anyidea wich 
one is better ?

I guess i will start with opensips, becasue old openser.org point to there.


Thanks again!
[]'sf.rique 



On Fri, Apr 29, 2011 at 10:49 AM, vip killa vipki...@gmail.com wrote:

could you send me book?

On Fri, Apr 29, 2011 at 9:48 AM, satish patel satish...@hotmail.com wrote:







I have sent you book in PM.

-S

Date: Fri, 29 Apr 2011 10:39:56 -0300
From: sf.ri...@gmail.com
To: asterisk-users@lists.digium.com


Subject: Re: [asterisk-users] Multiple Asterisk

Thanks!

Would apreciate the book!

But i am already researching
[]'sf.rique 



On Fri, Apr 29, 2011 at 10:10 AM, Satish Patel satish...@hotmail.com wrote:


Don't expect lots of thing because I have just post my basic config and method 
to integrate openser with asterisk and I did that 3 year ago.


http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration.




I would say search on google today lots of material are there and I have 
remembered there is a nice book regarding this. I guess I have PDF version of 
that book I will search and try to find.




--Sent from my iPhone
On Apr 29, 2011, at 8:40 AM, Henrique Fernandes sf.ri...@gmail.com wrote:




Can you post later t he link for it ?

I read alot that page.

[]'sf.rique 



On Fri, Apr 29, 2011 at 9:35 AM, Satish Patel satish...@hotmail.com wrote:




True, we had setup before openser with asterisk and it works great. I have 
wrote small document on voip-info related my project. 

--Sent from my iPhone


On Apr 29, 2011, at 8:23 AM, Henrique Fernandes sf.ri...@gmail.com wrote:





Thnaks a Lot.

So i will look for openser integration with asterisk!
[]'sf.rique 



On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote:





Hi,

If u want to setup for 4500 or more phone then better to user OpenSER + 
Asterisk.
OpenSER easily work for 10,000 calls.
You need to setup one server for OpenSER and all phone register on this server. 
You need to write routing logic in OpenSER server to call connect and if u need 
to play media then forward to destination asterisk server.






1 OpenSER server + Asterisk server for each location.

-- 
Best Regards,

Rajnikant Vanza
Software Engineer
---






Working On Linux,C/C++,VoIP,Asterisk Technology

On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes sf.ri...@gmail.com wrote:






No one ?

Other thing, i was reading about asterisk realtime, it can be configured to 
have multiple asterisk conectted to the same database? But how would it know in 
wich host are the number??







Thanks!

[]'sf.rique 



On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes sf.ri...@gmail.com wrote:







I am reading about, and some people are saying that openser is better for biger 
envoriments, and dundi is fine for smal envoriments, does anyone have any info 
about it ?

We have now about 4500 convencional phones and we gonna expand a lot.









So, 

OpenSER vs DUNDi ? 

I guess i will use Asterisk RealTime also right ?

[]'sf.rique 



On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham rizwanhas...@gmail.com wrote:









Here is a better link for DUNDi

http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/











skip the part which you know already

On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.com wrote:










[]'sf.rique 



On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger pabelan...@digium.com wrote:











On 11-03-15 06:19 PM, Henrique Fernandes wrote:


Have many diferenet locations that have convencional phones that need to

call others locations with convencional phones. And we can not change this,

I was reading and asterisk cannot handle it self this kind of setup, it

needs an separated serrver to control and routers the calls to this poins

right ?



So can you guys give any help ? I guess asterisk with SER could do the job ?




I don't believe SER will help you in the setup (see below).




So my question is how do i make the 2 PABX with asterisk talk to  each

other?  Do i need only 2 asterisk with digium or i need one server with SER

to maki it happen ? There is another program that does what i am looking for

?




If you require local hardware for each site, then you can install Asterisk at 
each location.  You can then interconnect 

[asterisk-users] SIP bad request

2011-04-29 Thread Mike
Hi,

 

I have been getting reports phones ringing only a tiny moment and then going
to voicemail.  CLI output shows:

 

-- SIP/user-0006fcdd is ringing

-- Got SIP response 400 Bad Request back from 23.23.23.23

-- SIP/user-0006fcdd is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)

 

Which does explain it.  How can I find the root cause of bad request?
Call-limit is very high for this sip user, so I`m not reaching that limit
for sure.

 

Mike

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

Re: [asterisk-users] SIP bad request

2011-04-29 Thread Захаров Антон

Try to look in 'sip set debug peer user'.

On 29.04.2011 18:10, Mike wrote:


Hi,

I have been getting reports phones ringing only a tiny moment and then 
going to voicemail.  CLI output shows:


-- SIP/user-0006fcdd is ringing

-- Got SIP response 400 Bad Request back from 23.23.23.23

-- SIP/user-0006fcdd is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)

Which does explain it.  How can I find the root cause of “bad 
request”? Call-limit is very high for this sip user, so I`m not 
reaching that limit for sure.


Mike


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

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


--
_
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   http://www.asterisk.org/hello

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   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Local channel scenario flushes CDR before dialplan end

2011-04-29 Thread Grigoriy Puzankin
Hi,

There's a quite complex dialplan scenario and I found out that CDR of
main channel is flushed right after hangup on Local channel. I will try
to simplify my scenario:

[incoming]
exten = 555,1,Noop(do something before using local channel, fill some
variables, play IVR menus and so on)
same = n,Dial(Local/555@office/n,,g)
same = n,Noop(Notice the option /n and flag g, which allows to
continue the dialplan after a destination channel hangs up, even it was
transfered by a connected peer - it is very important for me)
same = n,Noop(process some data, ask caller to value quality of service
- another IVR, record some messages)
same = n,Hangup()

exten = h,1,Noop(I'm using func_odbc to save quiz results into DB,
process recorded files, etc.)
same = n,Noop(I'm using cdr_adaptive to store custom fields in table
columns)
same = n,CDR(my_custom_field_a)=my_value
same = n,CDR(my_custom_field_z)=my_value

[office]
exten = 555,1,Dial(SIP/555)
same = n,Hangup()

A call comes from a SIP trunk directly to 555@incoming. It forks new
pair of Local channels, bridging other leg to SIP/555. SIP peer answers
the call, then hangs up. Dialplan continues right after Dial(Local/...).
Also it goes to h extension after reaching Hangup in 555@incoming.
Everything looks good, but CDR custom fields are empty, regardless that
verbose shows that they were set in dialplan. After a short
investigation I found out that CDR is written to DB in the same time
when dialplan exits Dial application. It produces to records: SIP trunk
to Local and Local to SIP/555, which is correct.

If I use SIP channel instead of Local, then CDR is written after
dialplan ends and all fields are set. But in this case I loose call
processing after it was transfered to another party (I have a lot of
contexts - catching a call-end is a pain).

Is it a bug or intended behavior?

Best regard,
Grigoriy.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] SIP bad request

2011-04-29 Thread Mike
What I am looking for?  Here is a snippet, with some info obfuscated. I can see 
the bad request, but why there is such a message isn’t obvious.

 

 

 

--- SIP read from UDP:23.23.23.23:23725 ---

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport

From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af

To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F

CSeq: 102 INVITE

Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66

Contact: sip:user@192.168.1.90:5060

User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734

Allow-Events: talk,hold,conference

Accept-Language: fr-fr,fr;q=0.9,en;q=0.8

Content-Length: 0

 

-

--- (11 headers 0 lines) ---

--- SIP read from UDP:23.23.23.23:23725 ---

SIP/2.0 400 Bad Request

Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport

From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af

To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F

CSeq: 102 INVITE

Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66

Contact: sip:user@192.168.1.90:5060

User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734

Accept-Language: fr-fr,fr;q=0.9,en;q=0.8

Content-Length: 0

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ??? ?
Sent: Friday, April 29, 2011 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP bad request

 

Try to look in 'sip set debug peer user'. 

On 29.04.2011 18:10, Mike wrote: 

Hi,

 

I have been getting reports phones ringing only a tiny moment and then going to 
voicemail.  CLI output shows:

 

-- SIP/user-0006fcdd is ringing

-- Got SIP response 400 Bad Request back from 23.23.23.23

-- SIP/user-0006fcdd is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)

 

Which does explain it.  How can I find the root cause of “bad request”? 
Call-limit is very high for this sip user, so I`m not reaching that limit for 
sure.

 

Mike

 
 
--
_
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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 --
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   http://www.asterisk.org/hello

asterisk-users mailing list
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-29 Thread satish patel

Hey Matt,

I have download irc linux base CLI client and connect to irc.freenode.net  i 
can see bunch or channels but i didn't find any #asterisk or #asterisk-bugs 
name. Am i looking at wrong place ?

*** #asterisk You're not on that channel
*** #asterisk Cannot join channel (+r) - you need to be identified with services

/JOIN #asterisk



 Date: Fri, 29 Apr 2011 14:26:46 +1200
 From: li...@venturevoip.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
 
 On 29/04/11 1:16 PM, Ira wrote:
  Well, I've no idea how to do that. I can duplicate the problem every
 
 IRC is an online chat system like MSN or Skype except that it's more 
 like a mailing list - you can talk to lots of people at the same time.
 
 On Windows you can use a program like mIRC to connect to 
 irc.freenode.net or even a plugin in Firefox.
 
 Once you're connected to IRC you can join chat rooms.
 
 There are some like #asterisk for discussion about Asterisk and 
 #asterisk-bugs for discussion about Asterisk bugs.
 
 Post back here if you have any problems connecting.
 
 -- 
 Cheers,
 
 Matt Riddell
 ___
 
 http://www.venturevoip.com/news.php (Daily Asterisk News)
 http://www.venturevoip.com/exchange.php (Full ITSP Solution)
 http://www.venturevoip.com/cc.php (Call Centre Solutions)
 
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-29 Thread Mark Deneen
Satish,

You must register your handle with freenode, because the asterisk
channel only allows registered people in.

http://freenode.net/faq.shtml#nicksetup

-M

On Fri, Apr 29, 2011 at 11:41 AM, satish patel satish...@hotmail.com wrote:
 Hey Matt,

 I have download irc linux base CLI client and connect to irc.freenode.net  i
 can see bunch or channels but i didn't find any #asterisk or #asterisk-bugs
 name. Am i looking at wrong place ?

 *** #asterisk You're not on that channel
 *** #asterisk Cannot join channel (+r) - you need to be identified with
 services

 /JOIN #asterisk



 Date: Fri, 29 Apr 2011 14:26:46 +1200
 From: li...@venturevoip.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4
 behind?

 On 29/04/11 1:16 PM, Ira wrote:
  Well, I've no idea how to do that. I can duplicate the problem every

 IRC is an online chat system like MSN or Skype except that it's more
 like a mailing list - you can talk to lots of people at the same time.

 On Windows you can use a program like mIRC to connect to
 irc.freenode.net or even a plugin in Firefox.

 Once you're connected to IRC you can join chat rooms.

 There are some like #asterisk for discussion about Asterisk and
 #asterisk-bugs for discussion about Asterisk bugs.

 Post back here if you have any problems connecting.

 --
 Cheers,

 Matt Riddell
 ___

 http://www.venturevoip.com/news.php (Daily Asterisk News)
 http://www.venturevoip.com/exchange.php (Full ITSP Solution)
 http://www.venturevoip.com/cc.php (Call Centre Solutions)

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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
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Re: [asterisk-users] odbc error - server is gone

2011-04-29 Thread Warren Selby
You're using 1.4.2. Why not try upgrading to a more recent release of 1.4 (I 
believe 1.4.41 is current) and see if your issue has been resolved. 

Thanks,
--Warren Selby, dCAP

On Apr 29, 2011, at 7:32 AM, Rizwan Hisham rizwanhas...@gmail.com wrote:

 Yes I have it there, here the content of the file:
 
 i think the code is buggy,
 
 here is a comment from the function which generated the error 
 (ast_odbc_smart_execute in res_odbc.c line 155 )
 
 /* This is a really bad method of trying to correct a dead connection.  It
  * only ever really worked with MySQL.  It will not work with any other
  * database, since most databases prepare their statements on the server,
  * and if you disconnect, you invalidate the statement handle.  Hence, if
  * you disconnect, you're going to fail anyway, whether you try to execute
  * a second time or not.
  */
 
 This function is used all over.
 
 On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan 
 sherwood.mcgo...@gmail.com wrote:
 On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.com 
 wrote:
 Hi list,
 yesterday I converted my voicemail.conf to realtime voicemail and also 
 configured to store the voicemessages in a database using odbc as described 
 here and here.
 I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver 
 for mysql on the server. I successfully completed the conversion of a lot of 
 voicemail users into db yesterday. But today on the CLI thsi error was 
 showing;
 
 [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL 
 Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 
 Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
 [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL 
 Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 
 Driver][mysqld-5.0.68-log]MySQL server has gone away (70)
 [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL 
 Execute error!
 [SELECT COUNT(*) FROM voicemessages WHERE dir = 
 '/var/spool/asterisk/voicemail/default/1757XXX/INBOX']
 
 I know that the error is caused due to stale odbc connection with mysql. But 
 i want to find out if there is a cure for it. Why the connection went stale 
 in the first place also.
 
 Any ideas?
 
 -- 
 Best Ragards
 Rizwan Qureshi
 VoIP/Asterisk Engineer
 Axvoice Inc.
 
 V: +92 (0)  6767 26
 E: rizwanhas...@gmail.com
 W: www.axvoice.com
 
 
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 do you have sanitysql = select 1 configured in res_odbc.ini?
 
 -- 
 Sherwood McGowan
 Telecommunications and VOIP Consultant
 
 
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 -- 
 Best Ragards
 Rizwan Qureshi
 VoIP/Asterisk Engineer
 Axvoice Inc.
 
 V: +92 (0)  6767 26
 E: rizwanhas...@gmail.com
 W: www.axvoice.com
 
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-29 Thread Richard Zheng
  Now imagine that 1.4 stays at only security level. For first case we
  have 2 options: upgrading for security reasons to last version but
  then no more voicemail, or staying with 1.4.26. In the second case,
  upgrading both servers to test with 1.8. If it's still not working, it
 was time
  loose beside other problems.

 If there are obvious regressions in major functionality such as voicemail,
 I'm more than happy to still consider making fixes for those problems during
 the security maintenance period.  It has to be pretty clear, though, and
 in this particular case, it is.

 Voicemail has been through several issues. Can't remember the details, we
experienced issues when imap was added. It broke the file based voicemails
even when imap was not used. As long as major bugs, like this and deadlocks
are taken care of during the 'security maintenance' period, most people are
happy. New features should be only added as a separate patch for risk
takers. The main branch should be major bugs only.
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

2011-04-29 Thread Danny Nicholas
To add another shilling to the pot - 

 

Asterisk as a whole and 1.4 specifically is a very good product.  Problems
are introduced (IMHO) when y'all take something that works perfectly well
and try to over-engineer it as a release bell-and-whistle instead of an
add-on.  Voicemail and Multi-tenant parking are great examples that come to
mind.  If you're a clunker like me, you would rather keep everything in text
files and use AGI's to do bell-and-whistle stuff.  But NOOO - you have to
disable these new features because the developer was so hot to get it into
the new release that he (that's a royal he) puts it up for inclusion and
we give 1000 users a bad impression of the new release when he could have
made it an add-on and more thoroughly tested it and gave us all that 3 days
of sleep we will never get back.

 

From a simple point of view, 1.8 works just as well as 1.4 for 99 percent of
everything I do, but then again, I don't do nearly as much as other posters
on here.

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[asterisk-users] HA Asterisk

2011-04-29 Thread Kaushal Shriyan
Hi,

I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf,
but its not yet production ready. Can someone please pitch in about HA
feature in Asterisk ?
(HA - High Availability.) Also, What would be the pros and cons of using
AsteriskNow over Asterisk ? Are the versions same in Asterisk and
AsteriskNow ? We have been evaluating Asterisk for our Voice Application and
it seems it would fit the requirement. Is Asterisk a CPU Intensive or a
Memory Intensive application.

Please suggest/guide.

Regards,

Kaushal
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Re: [asterisk-users] HA Asterisk

2011-04-29 Thread Michelle Dupuis
For the High Availability part check out the HAAST add-on for Asterisk at 
www.generationd.com 
It detects a variety of failures, shuts down the failing system, starts 
asterisk on the peer, moves the IP over, etc.  Runs with every Asterisk variant 
and every Linux distro.  No special hardware required.

Michelle
(I'd tell you how great it is but I work for Generation D ...)


From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan 
[kaushalshri...@gmail.com]
Sent: Friday, April 29, 2011 10:29 PM
To: Asterisk Users List
Subject: [asterisk-users] HA Asterisk

Hi,

I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but 
its not yet production ready. Can someone please pitch in about HA feature in 
Asterisk ?
(HA - High Availability.) Also, What would be the pros and cons of using 
AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? 
We have been evaluating Asterisk for our Voice Application and it seems it 
would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive 
application.

Please suggest/guide.

Regards,

Kaushal

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Re: [asterisk-users] Orginate not working well with PSTN lines

2011-04-29 Thread Ashik Ali
I thank everyone, for their fruitfull informations.

Regards,
Ashik Ali

On Fri, Apr 29, 2011 at 2:04 AM, Gilles codecompl...@free.fr wrote:
 On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
 beaasteriskg...@gmail.com wrote:
Anybody can explain me why asterisk is unable to detect ringback tone
from PSTN telco  ? .

 I guess it was a lot of work, and nobody bothered adding this to the
 Zaptel driver.


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