[asterisk-users] chan_ooh323 - cisco call manager express

[Dmitry Melekhov]

Hello!


I need to setup h323 trunk between cisco call manager express ( I have 
no access to it) and asterisk ( my side ).


Calls from asterisk are OK, but there is no voice if calls are from 
cisco to asterisk.


Looks like there is signalling problem.

Could you , please look at 
https://issues.asterisk.org/jira/browse/ASTERISK-27138


and give me any suggestions?


Thank you!



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Re: [asterisk-users] Pre-Dial Handler return something like GOSUB_RESULT?

[Richard Mudgett]

On Tue, Jul 18, 2017 at 6:49 PM, John Kiniston 
wrote:

> I'm messing around with pre-dialer handlers today and running into a wall.
>
> Dial has the U option where I can execute a Gosub when the channels bridge
> and there I can set the variable GOSUB_RESULT to BUSY to make Dial act like
> the channel I called was Busy.
>

> I want to do something similar with a Pre-Dial handler but don't see a way
> I can Set a variable or return a value that will cause Dial to act like the
> channel I called was Busy?
>
> Use case:
> Endpoint 100 calls Extension 101
>
> Extension 101 has a Pre-Dial Handler that checks how many calls Endpoint
> 101 has in progress and if it's greater than X returns a Busy.
>
> Dial acts like it got a Busy back from the Endpoint, Sets DIALSTATUS and
> continues through it's dial-plan.
>
> I've tried using the BUSY() Application inside my Pre-Dial handler.
> I've tried sending BUSY back as a Value with Return() to be picked up in
> GOSUB_RETVAL
> I've tried setting DIALSTATUS to BUSY.
>
> Am I trying to use the wrong tool for the Job here?
>

Why don't you do the how many calls the endpoint has check before Dial()?
You can use the LOCK/UNLOCK functions as shown in [1] on the calling
channel pre-dial
routine to prevent reentrancy issues while doing the check.  The called
channel pre-dial
routine is only to setup the channels you have decided to dial.


>
> Related, Why can we have multiple Hangup handlers but not Pre-Dial
> handlers?
>

* There is only one dial to execute the called channel pre-dial handler
while there are many opportunities to specify hangup handlers.
* How do you think you could associate different pre-dial handlers to
different called channels?

Richard

[1]
http://blogs.asterisk.org/2017/03/29/dialplan-handler-routines-allow-customization/
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[asterisk-users] Pre-Dial Handler return something like GOSUB_RESULT?

[John Kiniston]

I'm messing around with pre-dialer handlers today and running into a wall.

Dial has the U option where I can execute a Gosub when the channels bridge
and there I can set the variable GOSUB_RESULT to BUSY to make Dial act like
the channel I called was Busy.

I want to do something similar with a Pre-Dial handler but don't see a way
I can Set a variable or return a value that will cause Dial to act like the
channel I called was Busy?

Use case:
Endpoint 100 calls Extension 101

Extension 101 has a Pre-Dial Handler that checks how many calls Endpoint
101 has in progress and if it's greater than X returns a Busy.

Dial acts like it got a Busy back from the Endpoint, Sets DIALSTATUS and
continues through it's dial-plan.

I've tried using the BUSY() Application inside my Pre-Dial handler.
I've tried sending BUSY back as a Value with Return() to be picked up in
GOSUB_RETVAL
I've tried setting DIALSTATUS to BUSY.

Am I trying to use the wrong tool for the Job here?

Related, Why can we have multiple Hangup handlers but not Pre-Dial handlers?
-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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[asterisk-users] Asterisk 13.16.0 segfault

[Carlos Chavez]

I am getting frequent segfaults on a new Asterisk installation.  So far 
the only message I see is:


Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip 
7fb2d535723f sp 7fb25a11b5c0 error 4 in 
libasteriskpj.so.2[7fb2d52e5000+18]
Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip 
7f4afea0c23f sp 7f4a7f7e35c0 error 4 in 
libasteriskpj.so.2[7f4afe99a000+18]
Jul 18 09:22:57 pbxbogota kernel: asterisk[28471]: segfault at 188 ip 
7f2eb611923f sp 7f2e3aec25c0 error 4 in 
libasteriskpj.so.2[7f2eb60a7000+18]
Jul 18 09:25:49 pbxbogota kernel: asterisk[28949]: segfault at 188 ip 
7fc5758dd23f sp 7fc4fa6245c0 error 4 in 
libasteriskpj.so.2[7fc57586b000+18]
Jul 18 09:31:17 pbxbogota kernel: asterisk[29203]: segfault at 188 ip 
7f5f29abb23f sp 7f5eae8285c0 error 4 in 
libasteriskpj.so.2[7f5f29a49000+18]


Since this is a Freepbx distro does could the problem be related to 
their flavor of Asterisk?  I have several other plain Asterisk servers 
running on this version without any problems.  Any recommendations on 
how to debug this?


--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
dCAP #1349
+52 (55)8116-9161

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