similar to: Asterisk 1.8 SIP_CAUSE performance regression

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

Hello, I read on the wiki : Asterisk 1.8 will allow to read SIP response codes in the dialplan via *${HASH(SIP_CAUSE,<channel-name>)}*. Additionally make sure you're using the destination channel, not the source channel. But when I use this in my dialplan, this 'variable' is empty. Dialplan : exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})}) exten =>

Asterisk Project Security Advisory - AST-2011-001 Product Asterisk Summary Stack buffer overflow in SIP channel driver Nature of Advisory Exploitable Stack Buffer Overflow Susceptibility Remote Authenticated Sessions Severity Moderate

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

Asterisk Project Security Advisory - AST-2011-006 Product Asterisk Summary Asterisk Manager User Shell Access Nature of Advisory Permission Escalation Susceptibility Remote Authenticated Sessions Severity Minor

Hi, I am experimenting with getting hold of the sip cause and sip response from outgoing call. How could i make a userevent printing the sip cause and/or sip response. I have tried using hangupcause, sip_cause and such , but i am not getting any data. I would at least like to use the q.850 reason codes in the dialplan which i now am unable to do. Any help appreciated. [Beskrivning: Fogwise -

B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was

Hi, I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this: [outbound] Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier)) Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME: ${DIALEDTIME}

I think HANGUPCAUSE is channel agnostic. See: core show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE the correct channel. On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > It seems very weird to me

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -------------- next

I''m getting more and more "execution expired" as systems checking and hit puppetdb for the first time (switching from a mysql instance). The command queue isn''t long (1-5, if anything, all the time), and ym master itself seems to be dealing well enough. I have seen the collection time growing higher and higher though. This is a ~2K node deployment, and one of the few

so when I''m reading the pick axe book second edition I don''t see anything like the syntax you see people using in rails code. Specifically when you see. belongs_to :Person has_many :Phones etc these are methods on ActiveRecord right? Why is this invocation syntax never described in the Pick Axe book? I do see things like attr_reader :some_attribute etc but you

Hello, We have installed RHEL 5.2 Server ( i386 image ) on a machine with 64GB of RAM. When booting the default PAE kernel ( 2.6.18-92.el5PAE ) that is shipped with RHEL 5.2 installation the OS detects 64GB of RAM. But when booting the Xen kernel ( 2.6.18-92.el5xen ) the OS detects only 14GB of the total amount of available RAM. Did anyone encounter this problem before ? Is it a normal

Hello All, Does anyone know what is the current status of RHEL 5.2 x86_64 with respect to the maximum amount of RAM detected by the Xen 64 bit kernel (the one that shipped with Red Hat distribution ) In the following link https://www.redhat.com/docs/manuals/enterprise/RHEL-5-manual/en-US/RHEL510/Virtualization_Guide/ch-virt-hw-support.html it states that 32GB of RAM is the maximum amount of

Borders show it has to be ordered but Amazon says they have it but don''t show any reviews so what exactly is the true availability of the second edition? --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to