similar to: DEBUG: Cannot find variable 'XXX' ??

On Fri, 12 Jun 2015 11:49:05 -0700 John Kiniston <johnkiniston at gmail.com> wrote: > Try this for CHAN_SIP: > > same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer > same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the > mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we > have a mailbox defined log into it Perfect.

Hi guys, I just ran into a funny issue here. I'm trying to virtualize our asterisk pbx onto vmware esxi. Here's a quick glance of the system: * Ubuntu 9.10 i386 with linux-rt kernel (to get 1000Hz timer) everything up2date. * Asterisk 1.6.2.6 If I run asterisk using the debian init script in contrib/init.d/, top shows asterisk is using 99.x% CPU doing nothing. If I run asterisk with

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

I have this in my sip.conf: exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}") same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s) same => n,Hangup However, my extensions are set up so that they always show the external number, not the extension: [foobar2](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx callerid=Candace <5555551212>

You can use the following: switch3*CLI> show function SIPCHANINFO switch3*CLI> -= Info about function 'SIPCHANINFO' =- [Syntax] SIPCHANINFO(item) [Synopsis] Gets the specified SIP parameter from the current channel [Description] Valid items are: - peerip The IP address of the peer. - recvip The source IP address of the peer. - from

I changed the code to be more prototype-esque, and created a class called XMLDoc. I may add more functionality to it later, hence the more generic name, but you do something like this to convert XML to a hash: XMLDoc = Class.create(); Object.extend(XMLDoc.prototype, { initialize: function (xmlDoc) { this.element = xmlDoc; }, asHash: function () { if (! this._xmlHash) {

Hi, While compiling 1.8.8.1, I met the following error: [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o

I have one XML file with 30MB that I need to read the data. I try this; library(XML) doc <- xmlDoc("Malaria_Grave.xml") And R answers like this *** caught segfault *** address 0x5, cause 'memory not mapped' Traceback: 1: .Call("RS_XML_createDocFromNode", node, PACKAGE = "XML") 2: xmlDoc("Malaria_Grave.xml") Possible actions: 1: abort (with

Hi all, Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/${EXTEN}@user) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

No, the disable-xmldoc doesn't disable the whole doc creating procedure. Is there a way to disable it completely? Regarding the issue... Of course, I Can open a ticket, just I don't know about what exactly. I want to compile it without doc generate to make the asterisk module loads up fine. On Dec 4, 2016 8:41 PM, "Joshua Colp" <jcolp at digium.com> wrote: > On Sun,

I have a new Sangoma A102 and I'm trying to get it running in asterisk. A look through the dmesg log shows the card is detected and the various channels created. However, when I start asterisk I get the error below. Any ideas? My zapata.conf is below. Thanks, MD == Registered custom function SIPCHANINFO == Registered custom function CHECKSIPDOMAIN == Manager registered action

I have the attached class that I''m writing. The problem that I''m running into is that I can not access the options from the createArray function which get called after the ajax request gets done. I want to move the values of the xml file to an array and story it in the options variable. Can someone explain what I''m doing wrong? var LeaderInfo = Class.create();

Hi, I tried to run the make progdocs, but the first time, it said, I have no doxygen installed. So I compiled the latest release and reconfigure the asterisk. And after it, ut sucessfully started to build the docs. But it took a lot of time, So finally I aborted the process... Is there a way to disable doc creating? The --disable-xmldoc is enough? Thanks! On Dec 2, 2016 3:36 PM, "Joshua

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

Dear All, I'm installing the asterisk-1.6.2.24 in Centos 5.3, whenever i'm running following command ./configure I got below error: configure: *** XML documentation will not be available because the 'libxml2' development package is missing. configure: *** Please run the 'configure' script with the '--disable-xmldoc' parameter option configure: *** or install the

What is the current best practice to avoid transcoding on an outgoing call to a party whose codec preference is not known in advance? In other words, incoming calls are easy since codecs are negotiated from least-known (the remote party) to most-known (my endpoint) and my codecs can simply be preferred accordingly to match the remote. Outbound calls seem harder. Our endpoints always negotiate

Hello everybody, I want to use the TIMEOUT() function, but in the CLI the "show functions" command only shows 7 custom functions: QUEUEAGENTCOUNT SORT CUT CHECKSIPDOMAIN SIPCHANINFO SIPPEER SIPHEADER In addition, sometimes I get the debug message "function LANGUAGE not registered". How can I install those functions? I'm using Asterisk 1.2.10. Thanks in advance, --

Try this for CHAN_SIP: same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a mailbox defined log into it If you are using PJSIP it's more complex same => n,Set(EndPoint=${CHANNEL(endpoint)}) ; Get the peer same =>