Asterisk AMI SIP channel detect phone ringing

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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 Users 3.3 years ago 0 Answers

Asterisk 10.4.0 Now Available

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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) *…

Asterisk Users 3.3 years ago 0 Answers

Asterisk 1.8.12.0 Now Available

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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) *…

Asterisk Users 3.3 years ago 0 Answers

parsing issue

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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…

Asterisk Users 3.3 years ago 4 Answers

detecting intl. CLI with +

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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…

Asterisk Users 3.3 years ago 3 Answers

hangup problem on T1 span

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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 Users 3.3 years ago 1 Answer

Asterisk 8 and mixmonitor

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Hello, I have weird issue with Asterisk 8 lately. When I call MixMonitor without mixing the channels, it changes the
sides of "in" and "out".
Sometimes the first leg of the call is "in" and sometimes it's "out". I can't figure out if it's a known issue, or a new bug. I'm using Asterisk 8.11.1 Any ideas how can I figure out what is leg is what file ? Thanks,
Ido

Asterisk Users 3.3 years ago 3 Answers

CallerId back to incoming

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I'm currently doing some testing with Asterisk ( 1.8.11.0) on RHEL6
using realtime for sippeers, sipusers and musiconhold I have Avaya definity < -> PRI E1 < -> Asterisk 1 < -> IAX2 < -> Asterisk
2 I have peers (sip) snom 821s on both Asterisk 1 and 2 all calls working
between all systems. CallerID from Asterisk to Avaya is working correctly. The problem is a caller from Avaya to Asterisk displays correctly the
CID of the Asterisk Extension to the calling party on the Avaya but only
if the peer is on Asterisk…

Asterisk Users 3.3 years ago 0 Answers