Re: [asterisk-users] Asterisk Realtime Time Dial App

[Bruce Ferrell]

One thing I do is to use the mysql command to run simple queries and the
Goto a context with the information... simple and clean

On 09/25/2011 10:33 PM, Sam Govind wrote:
> Hmmm..interesting..I haven't came across anything like this so
> far..How about making a new table for the insertion of a new call
> data..and trigger some script to activate AMI/Call file according to
> new call data.
>
> http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10
>
> On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis  > wrote:
>
> Hello Everyone,
>
> I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
> wondering if it Is possible to have Asterisk make a calls based on a
> record inserted in a table realtime? If I have to develop
> something using AGI
> or AMI, I can do this  with a little direction?
>
> Thanks in Advance,
>
> Nick
>
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>
>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Ronald Cepres]

Hi Nick,

You mean if it is possible for Asterisk to use realtime dialplan? If it is,
AFAIK it is possible using a table format for realtime extensions.

Regards,
Ronald

On Mon, Sep 26, 2011 at 1:33 PM, Sam Govind  wrote:

> Hmmm..interesting..I haven't came across anything like this so far..How
> about making a new table for the insertion of a new call data..and trigger
> some script to activate AMI/Call file according to new call data.
>
>
> http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10
>
>
> On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis  wrote:
>
>> Hello Everyone,
>>
>> I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
>> wondering if it Is possible to have Asterisk make a calls based on a
>> record inserted in a table realtime? If I have to develop something using
>> AGI
>> or AMI, I can do this  with a little direction?
>>
>> Thanks in Advance,
>>
>> Nick
>>
>> --
>> _
>> -- 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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Re: [asterisk-users] bounty for ASTERISK-17474 streamingMusicOnHold bug

[Vladimir Mikhelson]

Great explanation.  Makes complete sense to me.

Any workaround you can think of?

-Vladimir



On 9/26/2011 12:26 AM, isr...@gmail.com wrote:
> It doesn't work at all with the dahdi timers 
> The reason it works it works till the first reload is because you are 
> preloading it before dahdi so it starts and uses the pthread timer later when 
> you reload it starts using the dahdi timer and there it goes 
>
>
> -Original Message-
> From: "Luke Hamburg" 
> Sender: asterisk-users-boun...@lists.digium.com
> Date: Mon, 26 Sep 2011 00:36:28 
> To: 'Asterisk Users Mailing List - Non-Commercial 
> Discussion'
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>   
> Subject: Re: [asterisk-users] bounty for ASTERISK-17474 streaming
>   MusicOnHold bug
>
> Danny Nicholas wrote:
>>> 2. Don't know if moving to 10.x would help you, but since that is still
> considered beta, that's probably not an option anyhow.
>
> Yup, not really an option for me.  I actually use this system daily and
> don't want to muck around with 10.0 just yet.
>
>>> 3. My understanding is that bounties need to be posted on the
> asterisk-dev list.
>
> Fair enough, I couldn't find that info - can anyone else confirm this?  I
> don't want to go barging into the dev list looking like a fool.
>
>>> 4. With those caveats, have you tried this: Copy the load_module and
> unload_module routines from res_timing_pthread.c to res_timing_dahdi.c
> (you'll probably need some includes [..]
>
> Hehe - no I definitely haven't tried that.  That's a bit above my pay grade
> for now.  I was hoping to find a more formal fix for this.  Still clinging
> onto the idea that with a decent bounty put together, someone who knows the
> code well enough would be able to fix this.  The fact that it WORKS GREAT
> until the first 'moh reload' suggests to me that it might be a relatively
> easy bug to squash.
>
>
>
>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Sam Govind]

Hmmm..interesting..I haven't came across anything like this so far..How
about making a new table for the insertion of a new call data..and trigger
some script to activate AMI/Call file according to new call data.

http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10

On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis  wrote:

> Hello Everyone,
>
> I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
> wondering if it Is possible to have Asterisk make a calls based on a
> record inserted in a table realtime? If I have to develop something using
> AGI
> or AMI, I can do this  with a little direction?
>
> Thanks in Advance,
>
> Nick
>
> --
> _
> -- 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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Re: [asterisk-users] bounty for ASTERISK-17474 streamingMusicOnHold bug

[isrlgb]

It doesn't work at all with the dahdi timers 
The reason it works it works till the first reload is because you are 
preloading it before dahdi so it starts and uses the pthread timer later when 
you reload it starts using the dahdi timer and there it goes 


-Original Message-
From: "Luke Hamburg" 
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 26 Sep 2011 00:36:28 
To: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] bounty for ASTERISK-17474 streaming
MusicOnHold bug

Danny Nicholas wrote:
>> 2. Don't know if moving to 10.x would help you, but since that is still
considered beta, that's probably not an option anyhow.

Yup, not really an option for me.  I actually use this system daily and
don't want to muck around with 10.0 just yet.

>> 3. My understanding is that bounties need to be posted on the
asterisk-dev list.

Fair enough, I couldn't find that info - can anyone else confirm this?  I
don't want to go barging into the dev list looking like a fool.

>> 4. With those caveats, have you tried this: Copy the load_module and
unload_module routines from res_timing_pthread.c to res_timing_dahdi.c
(you'll probably need some includes [..]

Hehe - no I definitely haven't tried that.  That's a bit above my pay grade
for now.  I was hoping to find a more formal fix for this.  Still clinging
onto the idea that with a decent bounty put together, someone who knows the
code well enough would be able to fix this.  The fact that it WORKS GREAT
until the first 'moh reload' suggests to me that it might be a relatively
easy bug to squash.




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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Steve Edwards]

On 09/25/2011 04:46 PM, jon pounder wrote:

Sometimes people get such swelled heads they need a slap back to 
reality - I completely agree with him the changes were idiotic.


Obviously the comments touched a nerve with you or you would not have 
replied.


On Sun, 25 Sep 2011, Alex Balashov wrote:

I don't think very highly of the changes either.  However, your approach 
and Bruce's is not how to make the case to the developers.


Bruce was a dickish troll and he was right. The Asterisk CLI was bad in 
1.2 and then veered into horrible.


Aside from that, is it really that big of a deal?  Is it that hard to 
learn a new command set and adapt?


Yes, it is.

I confess I'm a 1.2 Luddite so I have close to no experience with the 
current CLI. Every time I start using a newer version I get about 3 or 4 
commands into it and then I get sucked into the vortex of 'core or not 
core,' what module implements this command?, oh the hell with it -- I'd be 
done by now if I used 1.2.


Why should I have to know the name of the module before I can get help on 
a command? Which rocket scientist decided that some perfectly reasonable 
commands all of a sudden have to be prefaced with 'core' for no good 
reason? Which module is named core? How 'intuitive' is this? How 
'off-putting' is this to a new user? Why did previously maligned 
designer(s) decide to ignore every other reasonably* designed CLI and 
conclude that Asterisk's CLI must be 'different' and that obtuseness is a 
virtue?


Overcoming the inertia to retrain my ancient fingers is one of the reasons 
why I have not used a newer version in any of my new installs.


Yes, I could implement 'my own private Idaho' using CLI aliases but 
doesn't that seem like a lot of work and rather silly? I suspect I'd loose 
command completion in the process and I kind of like command completion.


I've ranted about this before and didn't get any traction so I'll crack 
open another beer and be quiet now.


*) 'Reasonably' is defined herein as 'as would be designed by a reasonable 
man**'


**) 'Reasonable man' is defined herein as 'me.'

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] bounty for ASTERISK-17474 streaming MusicOnHold bug

[Luke Hamburg]

Danny Nicholas wrote:
>> 2. Don't know if moving to 10.x would help you, but since that is still
considered beta, that's probably not an option anyhow.

Yup, not really an option for me.  I actually use this system daily and
don't want to muck around with 10.0 just yet.

>> 3. My understanding is that bounties need to be posted on the
asterisk-dev list.

Fair enough, I couldn't find that info - can anyone else confirm this?  I
don't want to go barging into the dev list looking like a fool.

>> 4. With those caveats, have you tried this: Copy the load_module and
unload_module routines from res_timing_pthread.c to res_timing_dahdi.c
(you'll probably need some includes [..]

Hehe - no I definitely haven't tried that.  That's a bit above my pay grade
for now.  I was hoping to find a more formal fix for this.  Still clinging
onto the idea that with a decent bounty put together, someone who knows the
code well enough would be able to fix this.  The fact that it WORKS GREAT
until the first 'moh reload' suggests to me that it might be a relatively
easy bug to squash.




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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Bruce B]

Thank you for a constructive reply. I am not a war monger and I appreciate a
proper response.

I will explore my options to that. My opinion may still be that such long
commands are unnecessary but at least it seems there is a way to go around
them for now and I am happy to hear that.



On Sun, Sep 25, 2011 at 9:23 PM, Paul Belanger wrote:

> On 11-09-25 08:57 PM, Bruce B wrote:
>
>> First of all, what the heck is this link you referenced:
>>
>>  http://lists.digium.com/pipermail/asterisk-users/2010-**
>> **April/247084.html
>> > 247084.html
>> >
>>
>>
>> ??
>>
>> Secondly, an Asterisk 1.6.2.18 that I am running right now plays nicely
>> with
>> "help" command. The 1.8 does do that. So, 1.6.2.18 has not been around for
>> 3
>> years. Again, stop misleading and changing the subject. When you state 3
>> years ago that is absolutely false. In doesn't apply to any of the
>> Asterisk
>> versions till 1.8xx
>>
>>  You seem to be missing the point or not reading my replies. The reason
> '*CLI> help' still works on asterisk 1.6.2, is because of the changes made 3
> years ago add res_clialiases.so.  Without it, the command would actually not
> work.
>
> Here is a simple test you can do on your 1.6.2 / 1.8 asterisk box:
>
> *CLI> module unload res_clialiases.so
> Unloaded res_clialiases.so
> *CLI> help
> No such command 'help' (type 'core show help help' for other possible
> commands)
>
> As you can see, without res_clialiases.so the command does not work. So, if
> you are saying the '*CLI> help' command does not work, then check your
> asterisk configuration first.
>
>
>  My post was very clear. Yes, it was sarcastic due to frustration but it
>> was
>> very clear and I wanted to say that there is no need to do "core show help
>> sip" when you can simply do "help sip".
>>
>> I still don't think your reply was called for. These trolls like I said
>> help
>> you live through with your attitude. If you were my employee and talked
>> like
>> this to anyone I would fire you right away.
>>
>> I am asking you nicely to please stop making this about yourself or
>> Digium.
>> Like I said, I like Asterisk. I love it. It works very good. Please listen
>> to the community feedback without getting so defensive. No one gains
>> anything from changes like this. I am sure Digium can afford one afternoon
>> meeting to decide what the commands naming convention should be for the
>> next
>> 20 years.
>>
>>  I don't even know how to reply to this, so I won't.  Thanks for all the
> fish.
>
>
> --
> 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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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Paul Belanger]

On 11-09-25 08:57 PM, Bruce B wrote:

First of all, what the heck is this link you referenced:

  
http://lists.digium.com/**pipermail/asterisk-users/2010-**April/247084.html

??

Secondly, an Asterisk 1.6.2.18 that I am running right now plays nicely with
"help" command. The 1.8 does do that. So, 1.6.2.18 has not been around for 3
years. Again, stop misleading and changing the subject. When you state 3
years ago that is absolutely false. In doesn't apply to any of the Asterisk
versions till 1.8xx

You seem to be missing the point or not reading my replies. The reason 
'*CLI> help' still works on asterisk 1.6.2, is because of the changes 
made 3 years ago add res_clialiases.so.  Without it, the command would 
actually not work.


Here is a simple test you can do on your 1.6.2 / 1.8 asterisk box:

*CLI> module unload res_clialiases.so
Unloaded res_clialiases.so
*CLI> help
No such command 'help' (type 'core show help help' for other possible 
commands)


As you can see, without res_clialiases.so the command does not work. 
So, if you are saying the '*CLI> help' command does not work, then check 
your asterisk configuration first.



My post was very clear. Yes, it was sarcastic due to frustration but it was
very clear and I wanted to say that there is no need to do "core show help
sip" when you can simply do "help sip".

I still don't think your reply was called for. These trolls like I said help
you live through with your attitude. If you were my employee and talked like
this to anyone I would fire you right away.

I am asking you nicely to please stop making this about yourself or Digium.
Like I said, I like Asterisk. I love it. It works very good. Please listen
to the community feedback without getting so defensive. No one gains
anything from changes like this. I am sure Digium can afford one afternoon
meeting to decide what the commands naming convention should be for the next
20 years.

I don't even know how to reply to this, so I won't.  Thanks for all the 
fish.


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

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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Bruce B]

First of all, what the heck is this link you referenced:

 
http://lists.digium.com/**pipermail/asterisk-users/2010-**April/247084.html

??

Secondly, an Asterisk 1.6.2.18 that I am running right now plays nicely with
"help" command. The 1.8 does do that. So, 1.6.2.18 has not been around for 3
years. Again, stop misleading and changing the subject. When you state 3
years ago that is absolutely false. In doesn't apply to any of the Asterisk
versions till 1.8xx

My post was very clear. Yes, it was sarcastic due to frustration but it was
very clear and I wanted to say that there is no need to do "core show help
sip" when you can simply do "help sip".

I still don't think your reply was called for. These trolls like I said help
you live through with your attitude. If you were my employee and talked like
this to anyone I would fire you right away.

I am asking you nicely to please stop making this about yourself or Digium.
Like I said, I like Asterisk. I love it. It works very good. Please listen
to the community feedback without getting so defensive. No one gains
anything from changes like this. I am sure Digium can afford one afternoon
meeting to decide what the commands naming convention should be for the next
20 years.


On Sun, Sep 25, 2011 at 4:05 PM, Paul Belanger wrote:

> On 11-09-25 02:23 PM, Bruce B wrote:
>
>> Paul,
>>
>> LOL...you are trying to change the subject. That's naive.
>>
>> You clearly know that I complained that there is no need for such drastic
>> changes and long commands. The fact that it's written in CHANGES file or
>> if
>> there was a commit for it doesn't make it any better. Stop with the flawed
>> reasoning.
>>
>> I am not going to complement your code or policies the whole time. Stop
>> wishing for that. I like Asterisk and I will raise a voice when I feel
>> uncomfortable with changes.
>>
>> All I am saying is that - Come up with a naming convention and for the
>> sake
>> of everyone stick to it. How hard could that be? Even with new features
>> you
>> can still stick to certain principles if you plan it ahead. If you don't
>> know how to do it, ask the community for input and people will help.
>>
>> -Bruce
>>
>>  You do realize this change happen almost 3 years go, aprox Nov. 2008.
> There was a discussion about it at Astricon, on -dev mailing list, plus a
> code review on reviewboard[1]. Implying it did not happen is incorrect.
>
> You might not have know about it because your first post from
> bruceb...@gmail.com seems to be Apr. 2010[2]. Community feedback was
> provided for the change, since it was driven by the community.
>
> If you don't like the change and want it reverted, simply load
> res_clialiases.so and edit cli_aliases.conf.
>
> Voicing your opinions is not a problem, however starting them with 'I don't
> mean to be rude but...' is not the best way to start them.  If you want to
> help shape the future of Asterisk, I encourage you to join the discussion on
> the asterisk-dev mailing lists.
>
> Its open source software, everybody gets a say.  It doesn't mean it will
> get done however.
>
> [1] 
> https://reviewboard.asterisk.**org/r/32/
> [2] http://lists.digium.com/**pipermail/asterisk-users/2010-**
> April/247084.html
>
>
> --
> 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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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[jon pounder]

On 09/25/2011 08:47 PM, Bruce B wrote:

This is becoming just like the bacula mailing list where anyone that 
knows anything is beaten into submission for daring to question the 
great and powerful oz.






You are very childish besides being very useless.

Also, note that there are others that are bothered by the same changes 
that are uncalled for. I was as constructive as possible but you think 
starting a sentence with "I am not trying to be rude..." is rude. LOL. 
I have said that upfront so idiots like you don't take offence but you 
did and you read as, "I am trying to be rude...". Well, suit yourself 
and keep sucking up Alex.




On Sun, Sep 25, 2011 at 4:41 PM, Alex Balashov 
mailto:abalas...@evaristesys.com>> wrote:


On 09/25/2011 02:23 PM, Bruce B wrote:

Stop wishing for that. I like Asterisk and I will raise a voice
when I feel uncomfortable with changes.


You won't get an audience if the way you go about it is dickish.

You're being a dick, and you know you're being a dick.  You're
just unwilling to admit it or intellectually engage with that.

If you were earnest and sincere about your desire to contribute
constructive criticism and effectuate change, you wouldn't start
the thread with a sarcastic subject line like "Who is the
'creative' mind behind changing Asterisk commands at CLI?"  That
has a mocking, derisive inflection, and you know it has a mocking,
derisive inflection.

If you expect to be taken seriously, you need to align your
behaviour with your stated objective--unless that's not actually
your objective, and in fact your objective is to be an
inflammatory jerk.

-- 
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] DID and how the caller id will appear

[C F]

Confirm with your provider that allow you to set caller id on outbound.


On Sun, Sep 25, 2011 at 1:59 PM, bilal ghayyad  wrote:
> Dear;
>
> By the way, the asterisk version is: 1.8.4.2
>
> Yes I tried Set(CALLERID(num)=5631040) as shown in the below dialing, and no 
> success. Also I tried Set(CALLERID(num)=1040) and I tried 
> Set(CALLERID(num)=065631040) as the city code is 06 and when we call any 
> mobile, it is appearing 065631040, but all of this did not work.
>
> Do I have to use SetCallerPres? What is the value?
>
> The E1 is located in Jordan and it is PRI with 30 channels. Is there any 
> thing need to be set other than Set(CALLERID(num)? I am afraid that I have to 
> set a specific value for SetCallerPress !
>
> Well, I have also a question: What should I set the callerid when I am 
> configuring the IP Phone in the sip.conf?
>
> By the way: what is the difference between using Set(CALLERID(num)=5631040) 
> and the callerid in the sip.conf?
>
> Kindly find below my dialing plan:
>
> exten => _90Z,1,Set(CALLERID(num)=5631040)
> exten => 
> _90Z,2,MixMonitor(${CHANNEL}${EXTEN:1}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}.wav)
> exten => _90Z,3,Dial(${PSTNTRUNK}/${EXTEN:1})
> exten => _90Z,4,Playback(vm-nobodyavail)
> exten => _90Z,5,Hangup()
> exten => _90Z,104,Congestion() ; if no channel available
> exten => _90Z,105,Hangup()
>
>
> ---
>
>
>>
>> Set(CALLERID(num)=5631040)
>> add this before the Dial command.
>>
>> On Sat, Sep 24, 2011 at 4:03 PM, bilal ghayyad 
>> wrote:
>> > Hi All;
>> >
>> > The DID range that we took from the telecom starts
>> from 1030 and end by 1059, now whenever we place a call, the
>> destination see the number 5631030. I gave the phone
>> extension 1040, and when I call, still the destination see
>> the number is 5631030?
>> >
>> > Kindly find below the configuration of the extension
>> 1040, please what I have also to configure so when this
>> extension make a call, the destination see it 5631040?
>> >
>> > [1040]
>> > type=friend
>> > host= dynamic
>> > callerid=1040
>> > disallow=all
>> > allow=alaw
>> > allow=ulaw
>> > allow=g729
>> > context=External
>> > dtmfmode=auto
>> > nat=no
>> > qualify=no
>> > canreinvite=yes
>> > username=1040
>> > secret=***
>> >
>> > Regards
>> > Bilal
>
>
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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Bruce B]

You are very childish besides being very useless.

Also, note that there are others that are bothered by the same changes that
are uncalled for. I was as constructive as possible but you think starting a
sentence with "I am not trying to be rude..." is rude. LOL. I have said that
upfront so idiots like you don't take offence but you did and you read as,
"I am trying to be rude...". Well, suit yourself and keep sucking up Alex.



On Sun, Sep 25, 2011 at 4:41 PM, Alex Balashov wrote:

> On 09/25/2011 02:23 PM, Bruce B wrote:
>
>  Stop wishing for that. I like Asterisk and I will raise a voice
>> when I feel uncomfortable with changes.
>>
>
> You won't get an audience if the way you go about it is dickish.
>
> You're being a dick, and you know you're being a dick.  You're just
> unwilling to admit it or intellectually engage with that.
>
> If you were earnest and sincere about your desire to contribute
> constructive criticism and effectuate change, you wouldn't start the thread
> with a sarcastic subject line like "Who is the 'creative' mind behind
> changing Asterisk commands at CLI?"  That has a mocking, derisive
> inflection, and you know it has a mocking, derisive inflection.
>
> If you expect to be taken seriously, you need to align your behaviour with
> your stated objective--unless that's not actually your objective, and in
> fact your objective is to be an inflammatory jerk.
>
> --
> 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/
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>  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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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Robert-iPhone]

I am adding dickish to my dictionary - thats a hot one!


Sent from my iPhone

On Sep 25, 2011, at 4:41 PM, Alex Balashov  wrote:

> On 09/25/2011 02:23 PM, Bruce B wrote:
> 
>> Stop wishing for that. I like Asterisk and I will raise a voice
>> when I feel uncomfortable with changes.
> 
> You won't get an audience if the way you go about it is dickish.
> 
> You're being a dick, and you know you're being a dick.  You're just unwilling 
> to admit it or intellectually engage with that.
> 
> If you were earnest and sincere about your desire to contribute constructive 
> criticism and effectuate change, you wouldn't start the thread with a 
> sarcastic subject line like "Who is the 'creative' mind behind changing 
> Asterisk commands at CLI?"  That has a mocking, derisive inflection, and you 
> know it has a mocking, derisive inflection.
> 
> If you expect to be taken seriously, you need to align your behaviour with 
> your stated objective--unless that's not actually your objective, and in fact 
> your objective is to be an inflammatory jerk.
> 
> -- 
> 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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>  http://www.asterisk.org/hello
> 
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[asterisk-users] Asterisk Realtime Time Dial App

[Nick Khamis]

Hello Everyone,

I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
wondering if it Is possible to have Asterisk make a calls based on a
record inserted in a table realtime? If I have to develop something using AGI
or AMI, I can do this  with a little direction?

Thanks in Advance,

Nick

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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[John Novack]

jon pounder wrote:

On 09/25/2011 04:41 PM, Alex Balashov wrote:

Sometimes people get such swelled heads they need a slap back to reality - I 
completely agree with him the changes were idiotic.

Obviously the comments touched a nerve with you or you would not have replied.




And let's not even THINK about mentioning the version number sequence changes!!!

Peg Leg O'Brien




On 09/25/2011 02:23 PM, Bruce B wrote:


Stop wishing for that. I like Asterisk and I will raise a voice
when I feel uncomfortable with changes.


You won't get an audience if the way you go about it is dickish.

You're being a dick, and you know you're being a dick.  You're just unwilling 
to admit it or intellectually engage with that.

If you were earnest and sincere about your desire to contribute constructive criticism 
and effectuate change, you wouldn't start the thread with a sarcastic subject line like 
"Who is the 'creative' mind behind changing Asterisk commands at CLI?"  That 
has a mocking, derisive inflection, and you know it has a mocking, derisive inflection.

If you expect to be taken seriously, you need to align your behaviour with your 
stated objective--unless that's not actually your objective, and in fact your 
objective is to be an inflammatory jerk.




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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Alex Balashov]

On 09/25/2011 04:46 PM, jon pounder wrote:


Sometimes people get such swelled heads they need a slap back to
reality - I completely agree with him the changes were idiotic.

Obviously the comments touched a nerve with you or you would not have
replied.


I don't think very highly of the changes either.  However, your 
approach and Bruce's is not how to make the case to the developers.


Aside from that, is it really that big of a deal?  Is it that hard to 
learn a new command set and adapt?


--
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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] Who is the "creative" mind behind changing Asterisk commands at CLI?

[jon pounder]

On 09/25/2011 04:41 PM, Alex Balashov wrote:

Sometimes people get such swelled heads they need a slap back to reality 
- I completely agree with him the changes were idiotic.


Obviously the comments touched a nerve with you or you would not have 
replied.






On 09/25/2011 02:23 PM, Bruce B wrote:


Stop wishing for that. I like Asterisk and I will raise a voice
when I feel uncomfortable with changes.


You won't get an audience if the way you go about it is dickish.

You're being a dick, and you know you're being a dick.  You're just 
unwilling to admit it or intellectually engage with that.


If you were earnest and sincere about your desire to contribute 
constructive criticism and effectuate change, you wouldn't start the 
thread with a sarcastic subject line like "Who is the 'creative' mind 
behind changing Asterisk commands at CLI?"  That has a mocking, 
derisive inflection, and you know it has a mocking, derisive inflection.


If you expect to be taken seriously, you need to align your behaviour 
with your stated objective--unless that's not actually your objective, 
and in fact your objective is to be an inflammatory jerk.





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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Alex Balashov]

On 09/25/2011 02:23 PM, Bruce B wrote:


Stop wishing for that. I like Asterisk and I will raise a voice
when I feel uncomfortable with changes.


You won't get an audience if the way you go about it is dickish.

You're being a dick, and you know you're being a dick.  You're just 
unwilling to admit it or intellectually engage with that.


If you were earnest and sincere about your desire to contribute 
constructive criticism and effectuate change, you wouldn't start the 
thread with a sarcastic subject line like "Who is the 'creative' mind 
behind changing Asterisk commands at CLI?"  That has a mocking, 
derisive inflection, and you know it has a mocking, derisive inflection.


If you expect to be taken seriously, you need to align your behaviour 
with your stated objective--unless that's not actually your objective, 
and in fact your objective is to be an inflammatory jerk.


--
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] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Paul Belanger]

On 11-09-25 02:23 PM, Bruce B wrote:

Paul,

LOL...you are trying to change the subject. That's naive.

You clearly know that I complained that there is no need for such drastic
changes and long commands. The fact that it's written in CHANGES file or if
there was a commit for it doesn't make it any better. Stop with the flawed
reasoning.

I am not going to complement your code or policies the whole time. Stop
wishing for that. I like Asterisk and I will raise a voice when I feel
uncomfortable with changes.

All I am saying is that - Come up with a naming convention and for the sake
of everyone stick to it. How hard could that be? Even with new features you
can still stick to certain principles if you plan it ahead. If you don't
know how to do it, ask the community for input and people will help.

-Bruce

You do realize this change happen almost 3 years go, aprox Nov. 2008. 
There was a discussion about it at Astricon, on -dev mailing list, plus 
a code review on reviewboard[1]. Implying it did not happen is incorrect.


You might not have know about it because your first post from 
bruceb...@gmail.com seems to be Apr. 2010[2]. Community feedback was 
provided for the change, since it was driven by the community.


If you don't like the change and want it reverted, simply load 
res_clialiases.so and edit cli_aliases.conf.


Voicing your opinions is not a problem, however starting them with 'I 
don't mean to be rude but...' is not the best way to start them.  If you 
want to help shape the future of Asterisk, I encourage you to join the 
discussion on the asterisk-dev mailing lists.


Its open source software, everybody gets a say.  It doesn't mean it will 
get done however.


[1] https://reviewboard.asterisk.org/r/32/
[2] http://lists.digium.com/pipermail/asterisk-users/2010-April/247084.html

--
Paul Belanger
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twitter: pabelanger | IRC: pabelanger (Freenode)
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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Bruce B]

Paul,

LOL...you are trying to change the subject. That's naive.

You clearly know that I complained that there is no need for such drastic
changes and long commands. The fact that it's written in CHANGES file or if
there was a commit for it doesn't make it any better. Stop with the flawed
reasoning.

I am not going to complement your code or policies the whole time. Stop
wishing for that. I like Asterisk and I will raise a voice when I feel
uncomfortable with changes.

All I am saying is that - Come up with a naming convention and for the sake
of everyone stick to it. How hard could that be? Even with new features you
can still stick to certain principles if you plan it ahead. If you don't
know how to do it, ask the community for input and people will help.

-Bruce





On Sun, Sep 25, 2011 at 1:22 PM, Paul Belanger wrote:

> On 11-09-25 01:01 PM, Bruce B wrote:
>
>> Paul,
>>
>> These trolls are the people who put your kid to school and put food on
>> your
>> table by giving valuable input and testing the open source software.
>>
>> Are you sure Digium endorses this stand of yours? Does everyone at Digium
>> think the users who gives feedback that is not exactly what you like is a
>> troll?
>>
>> WOW! I thought only rogue users try to censor this list but
>> congratulations
>> to Digium's own employees.
>>
>> Антон, Thanks. I will explore the option.
>>
>>  If you had bothered to search or even look at the CHANGES file, located
> in the source directory of asterisk, you would have seen the following:
>
>  * Cleanup another bunch of CLI commands. Now all modules follow the
>same schema. (Done by lmadsen, junky and mvanbaak during the devcon
>2008)
>
> Additionally, you could have taken the time to actually find the commit
> that made the change, since this is open source software everything is
> listed online [1].  Which was done by mvanbaak, an asterisk community
> member, not a Digium employee.
>
> [1] http://svnview.digium.com/svn/**asterisk?view=revision&**
> revision=145121
>
>
> --
> 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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Re: [asterisk-users] DID and how the caller id will appear

[bilal ghayyad]

Dear;

By the way, the asterisk version is: 1.8.4.2 

Yes I tried Set(CALLERID(num)=5631040) as shown in the below dialing, and no 
success. Also I tried Set(CALLERID(num)=1040) and I tried 
Set(CALLERID(num)=065631040) as the city code is 06 and when we call any 
mobile, it is appearing 065631040, but all of this did not work.

Do I have to use SetCallerPres? What is the value?

The E1 is located in Jordan and it is PRI with 30 channels. Is there any thing 
need to be set other than Set(CALLERID(num)? I am afraid that I have to set a 
specific value for SetCallerPress !

Well, I have also a question: What should I set the callerid when I am 
configuring the IP Phone in the sip.conf?

By the way: what is the difference between using Set(CALLERID(num)=5631040) and 
the callerid in the sip.conf?

Kindly find below my dialing plan:

exten => _90Z,1,Set(CALLERID(num)=5631040)
exten => 
_90Z,2,MixMonitor(${CHANNEL}${EXTEN:1}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}.wav)
exten => _90Z,3,Dial(${PSTNTRUNK}/${EXTEN:1})
exten => _90Z,4,Playback(vm-nobodyavail)
exten => _90Z,5,Hangup()
exten => _90Z,104,Congestion() ; if no channel available
exten => _90Z,105,Hangup()


---


> 
> Set(CALLERID(num)=5631040)
> add this before the Dial command.
> 
> On Sat, Sep 24, 2011 at 4:03 PM, bilal ghayyad 
> wrote:
> > Hi All;
> >
> > The DID range that we took from the telecom starts
> from 1030 and end by 1059, now whenever we place a call, the
> destination see the number 5631030. I gave the phone
> extension 1040, and when I call, still the destination see
> the number is 5631030?
> >
> > Kindly find below the configuration of the extension
> 1040, please what I have also to configure so when this
> extension make a call, the destination see it 5631040?
> >
> > [1040]
> > type=friend
> > host= dynamic
> > callerid=1040
> > disallow=all
> > allow=alaw
> > allow=ulaw
> > allow=g729
> > context=External
> > dtmfmode=auto
> > nat=no
> > qualify=no
> > canreinvite=yes
> > username=1040
> > secret=***
> >
> > Regards
> > Bilal


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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Alec Taylor]

We need explicit namespaces with asterisk CLI commands

On Mon, Sep 26, 2011 at 3:22 AM, Paul Belanger  wrote:
> On 11-09-25 01:01 PM, Bruce B wrote:
>>
>> Paul,
>>
>> These trolls are the people who put your kid to school and put food on
>> your
>> table by giving valuable input and testing the open source software.
>>
>> Are you sure Digium endorses this stand of yours? Does everyone at Digium
>> think the users who gives feedback that is not exactly what you like is a
>> troll?
>>
>> WOW! I thought only rogue users try to censor this list but
>> congratulations
>> to Digium's own employees.
>>
>> Антон, Thanks. I will explore the option.
>>
> If you had bothered to search or even look at the CHANGES file, located in
> the source directory of asterisk, you would have seen the following:
>
>  * Cleanup another bunch of CLI commands. Now all modules follow the
>    same schema. (Done by lmadsen, junky and mvanbaak during the devcon
>    2008)
>
> Additionally, you could have taken the time to actually find the commit that
> made the change, since this is open source software everything is listed
> online [1].  Which was done by mvanbaak, an asterisk community member, not a
> Digium employee.
>
> [1] http://svnview.digium.com/svn/asterisk?view=revision&revision=145121
>
> --
> 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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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Paul Belanger]

On 11-09-25 01:01 PM, Bruce B wrote:

Paul,

These trolls are the people who put your kid to school and put food on your
table by giving valuable input and testing the open source software.

Are you sure Digium endorses this stand of yours? Does everyone at Digium
think the users who gives feedback that is not exactly what you like is a
troll?

WOW! I thought only rogue users try to censor this list but congratulations
to Digium's own employees.

Антон, Thanks. I will explore the option.

If you had bothered to search or even look at the CHANGES file, located 
in the source directory of asterisk, you would have seen the following:


  * Cleanup another bunch of CLI commands. Now all modules follow the
same schema. (Done by lmadsen, junky and mvanbaak during the devcon
2008)

Additionally, you could have taken the time to actually find the commit 
that made the change, since this is open source software everything is 
listed online [1].  Which was done by mvanbaak, an asterisk community 
member, not a Digium employee.


[1] http://svnview.digium.com/svn/asterisk?view=revision&revision=145121

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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Steve Underwood]

On 09/26/2011 01:01 AM, Bruce B wrote:

Paul,

These trolls are the people who put your kid to school and put food on 
your table by giving valuable input and testing the open source software.


Are you sure Digium endorses this stand of yours? Does everyone at 
Digium think the users who gives feedback that is not exactly what you 
like is a troll?


WOW! I thought only rogue users try to censor this list but 
congratulations to Digium's own employees.
You must be new here. It is Digium's long term hostility to reasoned 
input that means very few of the early contributors to Asterisk still 
contribute today.


Steve




Антон, Thanks. I will explore the option.

-Bruce





On Sun, Sep 25, 2011 at 12:05 PM, Paul Belanger > wrote:


On 11-09-25 01:54 AM, Антон Квашёнкин wrote:

Just use cli aliases, provided by res_clialiases.so.

2011/9/25 Bruce Bmailto:bruceb...@gmail.com>>

Please don't feed the trolls. Thanks.

-- 
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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Bruce B]

Paul,

These trolls are the people who put your kid to school and put food on your
table by giving valuable input and testing the open source software.

Are you sure Digium endorses this stand of yours? Does everyone at Digium
think the users who gives feedback that is not exactly what you like is a
troll?

WOW! I thought only rogue users try to censor this list but congratulations
to Digium's own employees.

Антон, Thanks. I will explore the option.

-Bruce




On Sun, Sep 25, 2011 at 12:05 PM, Paul Belanger wrote:

> On 11-09-25 01:54 AM, Антон Квашёнкин wrote:
>
>> Just use cli aliases, provided by res_clialiases.so.
>>
>> 2011/9/25 Bruce B
>>
>>  Please don't feed the trolls. Thanks.
>
> --
> 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://lists.digium.com/**mailman/listinfo/asterisk-**users
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Re: [asterisk-users] Who is the "creative" mind behind changing Asterisk commands at CLI?

[Paul Belanger]

On 11-09-25 01:54 AM, Антон Квашёнкин wrote:

Just use cli aliases, provided by res_clialiases.so.

2011/9/25 Bruce B


Please don't feed the trolls. Thanks.

--
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twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server

[amit anand]

Hi
Do this script run without async option.

On Sun, Sep 25, 2011 at 18:19, Sam Govind  wrote:

> Oh! I was informed that Async:AGI is an AGI that is called in from AMI. Do
> tell more about it.
>
>
> On Sun, Sep 25, 2011 at 5:26 PM, Mehmet Avcioglu wrote:
>
>>
>> Actually it doesn't say "AGI(async:script)" it says "AGI(async:agi)" and
>> than continues further to setting up an AMI user so the script is executed
>> through the manager interface?? Than it says "AGI(agi:async)".?? Well most
>> importantly it says "Cons of async AGI: It is the most complex method of
>> using AGI to implement." ..:) I have been interested in Async AGI as well
>> and after reading your post looked into the link you provided, seems
>> different than what we immediately think, a background process.
>>
>> Perhaps just start the script normally "AGI(script.sh)" and than inside it
>> run your background process "background-script.sh > /dev/null 2>&1 <
>> /dev/null &" or fork a new process, detach, run in background, etc...
>>
>> Hopefully somebody else can point us towards the right direction in
>> setting up a real asterisk asynchronous AGI application.
>>
>> --
>> Mehmet
>>
>> On Sep 25, 2011, at 2:00 AM, Randall Degges wrote:
>>
>> > Hi Everyone,
>> >
>> > I've been trying to get asynchronous AGIs working in some Asterisk code
>> I have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and
>> AGI scripting overall. Here's my problem: I can't get Asterisk to execute
>> *any* AGIs asynchronously.
>> >
>> > Firstly, I discovered asynchronous AGIs via "Asterisk: The Definitive
>> Guide". The asynchronous AGI information I read can be found online, here:
>> http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the
>> section titled "Async AGI--AMI Controlled AGI").
>> >
>> > According to the book, since Asterisk 1.6.0 the AGI dialplan application
>> has been able to execute AGI scripts asynchronously, via the syntax:
>> >
>> > exten => s,1,AGI(async:script)
>> >
>> > According to the book, using the "async:" prefix should have Asterisk
>> run the AGI script in the background and instantly continue executing
>> dialplan code.
>> >
>> > So here's my Asterisk dialplan code that's being run:
>> >
>> > [hangup]
>> > exten => s,1,AGI(async:/etc/asterisk/scripts/hangup.py)
>> > exten => s,n,Return()
>> >
>> > Pretty simple context--essentially my AGI script just does some call
>> clean up logic before a caller hangs up, talking to a few web servers and
>> generating statistics for later usage. What happens when Asterisk executes
>> this context, is:
>> >
>> > WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute
>> '/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does
>> not exist.
>> >
>> > As you can see, Asterisk is ignoring the async: directive, and treating
>> it as part of the AGI script path.
>> >
>> > Is there anyway for me to make asynchronous AGIs work? I've tried
>> searching online to no avail.
>> >
>> > I'd greatly appreciate any responses, thanks for your time.
>> >
>> > -Randall
>> >
>> > --
>> > Randall Degges
>> > http://rdegges.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
>>
>>
>> --
>> _
>> -- 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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Re: [asterisk-users] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server

[Sam Govind]

Oh! I was informed that Async:AGI is an AGI that is called in from AMI. Do
tell more about it.

On Sun, Sep 25, 2011 at 5:26 PM, Mehmet Avcioglu wrote:

>
> Actually it doesn't say "AGI(async:script)" it says "AGI(async:agi)" and
> than continues further to setting up an AMI user so the script is executed
> through the manager interface?? Than it says "AGI(agi:async)".?? Well most
> importantly it says "Cons of async AGI: It is the most complex method of
> using AGI to implement." ..:) I have been interested in Async AGI as well
> and after reading your post looked into the link you provided, seems
> different than what we immediately think, a background process.
>
> Perhaps just start the script normally "AGI(script.sh)" and than inside it
> run your background process "background-script.sh > /dev/null 2>&1 <
> /dev/null &" or fork a new process, detach, run in background, etc...
>
> Hopefully somebody else can point us towards the right direction in setting
> up a real asterisk asynchronous AGI application.
>
> --
> Mehmet
>
> On Sep 25, 2011, at 2:00 AM, Randall Degges wrote:
>
> > Hi Everyone,
> >
> > I've been trying to get asynchronous AGIs working in some Asterisk code I
> have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and
> AGI scripting overall. Here's my problem: I can't get Asterisk to execute
> *any* AGIs asynchronously.
> >
> > Firstly, I discovered asynchronous AGIs via "Asterisk: The Definitive
> Guide". The asynchronous AGI information I read can be found online, here:
> http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the
> section titled "Async AGI--AMI Controlled AGI").
> >
> > According to the book, since Asterisk 1.6.0 the AGI dialplan application
> has been able to execute AGI scripts asynchronously, via the syntax:
> >
> > exten => s,1,AGI(async:script)
> >
> > According to the book, using the "async:" prefix should have Asterisk run
> the AGI script in the background and instantly continue executing dialplan
> code.
> >
> > So here's my Asterisk dialplan code that's being run:
> >
> > [hangup]
> > exten => s,1,AGI(async:/etc/asterisk/scripts/hangup.py)
> > exten => s,n,Return()
> >
> > Pretty simple context--essentially my AGI script just does some call
> clean up logic before a caller hangs up, talking to a few web servers and
> generating statistics for later usage. What happens when Asterisk executes
> this context, is:
> >
> > WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute
> '/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does
> not exist.
> >
> > As you can see, Asterisk is ignoring the async: directive, and treating
> it as part of the AGI script path.
> >
> > Is there anyway for me to make asynchronous AGIs work? I've tried
> searching online to no avail.
> >
> > I'd greatly appreciate any responses, thanks for your time.
> >
> > -Randall
> >
> > --
> > Randall Degges
> > http://rdegges.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
>
>
> --
> _
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] DID and the Caller ID for outgoing

[C F]

Did you add the Set(CALLERID(num) as I have previously pointed out?

On 9/25/11, bilal ghayyad  wrote:
> Hi All;
>
> I do not know if the SEP_Mac_address.cnf.xml of the cisco file is also has
> effecting on the DID and Caller ID to appear at the destination, because I
> found the following:
>
>
>  true
>  true
>  2
>  2
>
> So, does the 2 is effecting on
> displaying the caller id at the destination?
>
> What does it mean the value to be 2?
>
> Because I am placing the callerid=<5631040> (and I tried callerid=<5631040>
> also) in the sip.conf for the Cisco IP Phone, and no success, it is always
> displaying the primary number which is: 5631030
>
> So, I started think if the 2 at this
> setting file is effecting? Or if there is any parameter is effecting?
>
> Any help?
>
> Regards
> Bilal
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server

[Mehmet Avcioglu]

Actually it doesn't say "AGI(async:script)" it says "AGI(async:agi)" and than 
continues further to setting up an AMI user so the script is executed through 
the manager interface?? Than it says "AGI(agi:async)".?? Well most importantly 
it says "Cons of async AGI: It is the most complex method of using AGI to 
implement." ..:) I have been interested in Async AGI as well and after reading 
your post looked into the link you provided, seems different than what we 
immediately think, a background process.

Perhaps just start the script normally "AGI(script.sh)" and than inside it run 
your background process "background-script.sh > /dev/null 2>&1 < /dev/null &" 
or fork a new process, detach, run in background, etc...

Hopefully somebody else can point us towards the right direction in setting up 
a real asterisk asynchronous AGI application.

--
Mehmet

On Sep 25, 2011, at 2:00 AM, Randall Degges wrote:

> Hi Everyone,
> 
> I've been trying to get asynchronous AGIs working in some Asterisk code I 
> have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and AGI 
> scripting overall. Here's my problem: I can't get Asterisk to execute *any* 
> AGIs asynchronously.
> 
> Firstly, I discovered asynchronous AGIs via "Asterisk: The Definitive Guide". 
> The asynchronous AGI information I read can be found online, here: 
> http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the 
> section titled "Async AGI--AMI Controlled AGI").
> 
> According to the book, since Asterisk 1.6.0 the AGI dialplan application has 
> been able to execute AGI scripts asynchronously, via the syntax:
> 
> exten => s,1,AGI(async:script)
> 
> According to the book, using the "async:" prefix should have Asterisk run the 
> AGI script in the background and instantly continue executing dialplan code.
> 
> So here's my Asterisk dialplan code that's being run:
> 
> [hangup]
> exten => s,1,AGI(async:/etc/asterisk/scripts/hangup.py)
> exten => s,n,Return()
> 
> Pretty simple context--essentially my AGI script just does some call clean up 
> logic before a caller hangs up, talking to a few web servers and generating 
> statistics for later usage. What happens when Asterisk executes this context, 
> is:
> 
> WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute 
> '/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does 
> not exist.
> 
> As you can see, Asterisk is ignoring the async: directive, and treating it as 
> part of the AGI script path.
> 
> Is there anyway for me to make asynchronous AGIs work? I've tried searching 
> online to no avail.
> 
> I'd greatly appreciate any responses, thanks for your time.
> 
> -Randall
> 
> -- 
> Randall Degges
> http://rdegges.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


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[asterisk-users] DID and the Caller ID for outgoing

[bilal ghayyad]

Hi All;

I do not know if the SEP_Mac_address.cnf.xml of the cisco file is also has 
effecting on the DID and Caller ID to appear at the destination, because I 
found the following:


 true
 true
 2
 2

So, does the 2 is effecting on displaying 
the caller id at the destination? 

What does it mean the value to be 2?

Because I am placing the callerid=<5631040> (and I tried callerid=<5631040> 
also) in the sip.conf for the Cisco IP Phone, and no success, it is always 
displaying the primary number which is: 5631030

So, I started think if the 2 at this 
setting file is effecting? Or if there is any parameter is effecting?

Any help?

Regards
Bilal



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