Re: [asterisk-users] HA Asterisk

[RAJNIKANT VANZA]

Hi Kaushal,

I have done HA for Asterisk servers as well as SIP Server (kamailio).

Please write your detail requirement.

-> how many Asterisk Sever require for HA?
-> How much down time acceptable during Asterisk Sever failover?
-> Which type Asterisk Sever Failover u required?

Send me your detail requirement and answer of above question ASAP.

-- 
Best Regards,

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


On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan
wrote:

> 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

[Alex Balashov]

On 04/30/2011 02:13 AM, RAJNIKANT VANZA wrote:


Hi Kaushal,

I have done HA for Asterisk servers as well as SIP Server (kamailio).

Please write your detail requirement.

-> how many Asterisk Sever require for HA?
-> How much down time acceptable during Asterisk Sever failover?
-> Which type Asterisk Sever Failover u required?

Send me your detail requirement and answer of above question ASAP.


Requests for additional details are a lot more persuasive when 
conveyed in a semi-literate manner.


--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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

[Ashik Ali]

I thank everyone, for their fruitfull informations.

Regards,
Ashik Ali

On Fri, Apr 29, 2011 at 2:04 AM, Gilles  wrote:
> On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
>  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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Re: [asterisk-users] HA Asterisk

[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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[asterisk-users] HA Asterisk

[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] Discussion: Are we ready to leave 1.4 behind?

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

[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] odbc error - server is gone

[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  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 
>  wrote:
> On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham  
> 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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> 
> 
> -- 
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> 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?

[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  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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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[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] SIP bad request

[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" ;tag=as40e0c5af

To: "user" ;tag=372AEEC-62912E9F

CSeq: 102 INVITE

Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66

Contact: 

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" ;tag=as40e0c5af

To: "user" ;tag=372AEEC-62912E9F

CSeq: 102 INVITE

Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66

Contact: 

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

[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.

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Re: [asterisk-users] SIP bad request

[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


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[asterisk-users] SIP bad request

[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

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Re: [asterisk-users] Multiple Asterisk

[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  wrote:

could you send me book?

On Fri, Apr 29, 2011 at 9:48 AM, satish patel  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  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  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  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  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  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  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  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  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  wrote:










[]'sf.rique 



On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger  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 a

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

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

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

Re: [asterisk-users] Multiple Asterisk

[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  wrote:

> could you send me book?
>
>
> On Fri, Apr 29, 2011 at 9:48 AM, satish patel 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 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 
>> 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 

Re: [asterisk-users] Multiple Asterisk

[vip killa]

could you send me book?

On Fri, Apr 29, 2011 at 9:48 AM, satish patel  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 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 
> 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 bett

Re: [asterisk-users] Multiple Asterisk

[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  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  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  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  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  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  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  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  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  wrote:








[]'sf.rique 



On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger  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






--

_

-- Bandwidth and Col

Re: [asterisk-users] Multiple Asterisk

[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 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 
> 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 

Re: [asterisk-users] Multiple Asterisk

[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   
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  
 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   
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 > 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 > 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 > 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 > 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 > wrote:


[]'sf.rique


On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger  
 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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 http://www.asterisk.org/hello

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

V: +92 (0)  6767 26
E: rizwanh

Re: [asterisk-users] (no subject)

[Muhammad Usman]

you running GSM FWTs with asterisk ?

On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem wrote:

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



-- 
Regards:
(Muhammad υѕмαη )
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[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] Multiple Asterisk

[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  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 
> 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:
>>>   

Re: [asterisk-users] Multiple Asterisk

[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   
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  
 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 > 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 > 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  
 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 > wrote:


[]'sf.rique


On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger  
 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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--
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

[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 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
>>
>>
>> --
>> _
>> -- 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
>>
>
>
> do you have "sanitysql => select 1" configured in res_odbc.ini?
>
> --
> Sherwood McGowan
> Telecommunications and VOIP Consultant
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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

[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 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 
> 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 
>> 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 
>>> 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 >>> > wrote:

>
> []'sf.rique
>
>
> On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger 
> 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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>
>
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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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[asterisk-users] Hardware Server Configuration/8 or 4 port PRI Card

[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] Discussion: 1.8 quality issues

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

[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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