* You are viewing the archive for May 2nd, 2012

Asterisk AMI SIP channel detect phone ringing

Hey guys,

I am using a SIP trunk to make outgoing calls. Outgoing calls are
going through okay. I am using the AMI to Originate a call. The
channel is not returning any event when the phone on the PSTN is
ringing. How can i detect the phone ringing on the SIP channel?

Am desperate.

Thanks.

Asterisk 10.4.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 10.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 10.4.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* — Prevent chanspy from binding to zombie channels
(Closes issue ASTERISK-19493. Reported by lvl)

* — Fix Dial m and r options and forked calls generating warnings
for voice frames.
(Closes issue ASTERISK-16901. Reported by Chris Gentle)

* — Remove ISDN hold restriction for non-bridged calls.
(Closes issue ASTERISK-19388. Reported by Birger Harzenetter)

* — Fix copying of CDR(accountcode) to local channels.
(Closes issue ASTERISK-19384. Reported by jamicque)

* — Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
(Closes issue ASTERISK-19303. Reported by Jon Tsiros)

* — Eliminate double close of file descriptor in manager.c
(Closes issue ASTERISK-18453. Reported by Jaco Kroon)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.0

Thank you for your continued support of Asterisk!

Asterisk 1.8.12.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.12.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.12.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* — Prevent chanspy from binding to zombie channels
(Closes issue ASTERISK-19493. Reported by lvl)

* — Fix Dial m and r options and forked calls generating warnings
for voice frames.
(Closes issue ASTERISK-16901. Reported by Chris Gentle)

* — Remove ISDN hold restriction for non-bridged calls.
(Closes issue ASTERISK-19388. Reported by Birger Harzenetter)

* — Fix copying of CDR(accountcode) to local channels.
(Closes issue ASTERISK-19384. Reported by jamicque)

* — Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
(Closes issue ASTERISK-19303. Reported by Jon Tsiros)

* — Eliminate double close of file descriptor in manager.c
(Closes issue ASTERISK-18453. Reported by Jaco Kroon)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.12.0

Thank you for your continued support of Asterisk!

parsing issue

I get an error when I execute this code
exten => rejected,n,Hangup($[-1*${Z}])

May 2 13:42:09] WARNING[23128]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected $end, expecting
‘-‘ or ‘!’ or ‘(‘ or ‘‘; Input:
-1*

The variable “Z” has a negative number, which is the code that I need
to use in the hangup.
Any idea how can I do this? There is no ABS() function in Asterisk. I
already filed a request for it but it turns up that it will cost me
money. How can I remove the sign from a number?
Philip

detecting intl. CLI with +

Hello asterisk users,

I need to convert the CLI received according to national/international
format:

55-555-5555 to 055-555-5555 (add 0 in the beginning)
+55-55-555-5555 to +55-55-555-5555 (remains unchanged)

I put the following line in my dial plan:
exten => _X., n, Set(CALLERID(num)=${IF($[ ${CALLERID(num):0:1} =
"+"]?${CALLERID(num)}:0${CALLERID(num)})})

But I get these error messages:
[May 2 17:05:43] WARNING[1494]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected ‘+’, expecting $end;
Input:
+ = “+”
^
[May 2 17:05:43] WARNING[1494]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
[May 2 17:05:43] WARNING[1494]: func_logic.c:192 acf_if: Syntax
IF(?[
][:]) (expr must be non-null, and either
or
must be non-null)
[May 2 17:05:43] WARNING[1494]: func_logic.c:193 acf_if: In this
case, =”,
=’+55555555555′, and =’0+55555555555′

Can anyone suggest the proper syntax? I tried the + with no quotes, single
quotes ‘+’ and double quotes”+” and nothing worked.

Thanks

hangup problem on T1 span

Hello all,

I’m trying to solve a problem on a T1 span setup wherein calls are
apparently not hanging up properly.

The system in question is using a Xorcom Astribank with 1 full and 1
partial T1 span, and running Asterisk 1.4.36.

The symptom is that when a call hangs up on a DAHDI channel (according to
Asterisk), and another outgoing call tries to open a new channel on the
same line as the hung-up call within approximately a minute of the hangup,
the new call gets a congestion notice (“all circuits busy”) from
asterisk. After about a minute passes after the hangup, the line becomes
available again. So it seems like the channels are not hanging up when
Asterisk tells them to, and Asterisk doesn’t know it.

I suspected a signaling issue, and this appeared confirmed when I
discovered that the signalling was set in chan_dahdi.conf as “fxs_ks” (this
installation had been converted from analog lines by another company; I
guess that was an oversight?).

So I changed it to pri_cpe, as my reading of the docs indicated was proper.
After this change and restarting everything, though, the symptoms persist.
So I figure that either my reading of the docs is wrong (and therefore
pri_cpe is not the right signaling) OR something totally unrelated is going
on.

Can someone please clue me in here? I am a bit at a loss. Let me know if
you need further information about the system/environment.

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729