search for: kjcsb

The variable ${CONTEXT} stores the value of the current context. However if we are in a macro that will be the name of the macro. How do I access the name of the local channel's context. For example: [macro-test] exten => s,n,NoOp(Context ${CONTEXT}) CLI shows: -- Executing NoOp("Local/2592@1100006-2000-e802,2", "Context macro-test") in new stack I want to get

...t sure that any solution with the MySQL dialplan command is going to be ideal. You also can't nest your queries, ie the connectid/result id seems to only be good for one resultset at a time... try doing something like findme/followme with that! Doug. > -----Original Message----- > From: kjcsb [mailto:kjcsb@orcon.net.nz] > Sent: Monday, December 18, 2006 11:33 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Re: Best way to access MySQL data from dial > plan > > > Resending as message didn't show up the first time &gt...

----- Original Message ----- From: "kjcsb" <kjcsb@orcon.net.nz> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, January 24, 2007 8:24 AM Subject: Re: [asterisk-users] realtime sipusers and rtcachefriends... bigheadache!! > >> hi folks,...

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

...rn NULL; > } > <snip/> > } > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf > Of Ricardo > > Carvalho > > Sent: 20 October 2006 17:51 > > To: kjcsb; Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] SIP_HEADER function; what names are > > available? > > > > Any news on this thread? I also need to know the way to get > the R-URI > > from sip INVITE messages received by...

On attempting to make Zaptel 1.2.16 on FC5, I get the following messages: /usr/src/zaptel-1.2.16/xpp/xbus-core.c: In function 'debugfs_open': /usr/src/zaptel-1.2.16/xpp/xbus-core.c:171: error: 'struct inode' has no member named 'u' make[3]: *** [/usr/src/zaptel-1.2.16/xpp/xbus-core.o] Error 1 make[2]: *** [/usr/src/zaptel-1.2.16/xpp] Error 2 make[1]: ***

I have the following in the dialplan: [macro-systemrecording] exten => s,1,Goto(${ARG1},1) exten => dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav) exten => dorecord,n,Wait(1) exten => dorecord,n,Goto(confmenu,1) exten => docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording) exten => docheck,n,Wait(1) exten => docheck,n,Goto(confmenu,1) exten =>

I have had a TDM400 with 2 FXO and 2 FXS working for ages (>12 months). It has stopped working. All four green lights are still lit. I have rebuilt zaptel and asterisk and restarted but the problem persists. /sbin/ztcfg -vvvv Zaptel Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03:

I am trying to get to the bottom of audio clicks, pops, dropouts with my Asterisk server. These occur even when the system is under minimal load (e.g. 1 Zap device in a queue being played music on hold) and occurs with both Zap and Sip devices so isn't network related. The audio problems occur at the same time on all channels and seems to be when Asterisk "gets busy" and uses

>I have Asterisk listening for sip traffic on port 5060. I want to allow >users to use either port 80 or 5060 if they want. Hopefully this will avoid >some firewall issues. > > Is this a sensible/crazy thing to do? I have done a bunch of searching and > believe iptables can help but haven't been able to find an example to > forward something from 80 to 5060 inbound and

Resending as message didn't show up the first time >I need to access MySQL from the dial plan. Currently I am using the MYSQL >function: > exten => *78,n,MYSQL(Connect asterisklocal localhost asteriskuser password > asterisk) > exten => *78,n,MYSQL(Query resultid ${asterisklocal} CALL\ > sp_ins_into_avp(\'/DND/${CALLERID(number)}\'\,\'YES\')) >

>> The call that gets dropped had a retransmission of INVITE from UAC >> to UAS (and therefore retransmission of 200 OK from UAS to UAC). >> There is nothing wrong with the re-transmission as such, but I >> noticed a potential bug in Asterisk in the way it responds to an >> INVITE retransmission. Asterisk is bumping up the session version >> number in

...0.14.camel@localhost> Content-Type: text/plain; charset=ISO-8859-1 Hello. Take a look about function SIPPEER (asterisk -rx "show function SIPPEER"). It helps how to use peer information. Regards. Josi Luis El lun, 26-02-2007 a las 23:09 -0800, Yuan LIU escribis: > >From: kjcsb <kjcsb@yahoo.com> > >Date: Mon, 26 Feb 2007 22:32:29 -0800 (PST) > > > > >> CLI shows: > > >> -- Executing NoOp("Local/2592@1100006-2000-e802,2", "Context > >macro-test") in new stack > > >> > > >> I wa...

I am (unsuccessfully) trying to install zaptel (incl ztdummy - I don't have any Digium hardware) on CentOS 4. uname -r 2.6.9-42.ELsmp Not sure how this relates to 2.6.9-42.0.3 (see below) ln -s /usr/src/kernels/`uname -r` /usr/src/linux ln -s /usr/src/kernels/`uname -r` /usr/src/linux-2.6 cd /usr/src/zaptel-1.2.11* make linux26 You do not appear to have the sources for the 2.6.9-42.ELsmp

Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. ----- Mike Hammett Intelligent Computing

We wish to do the following: 1. receive an email with an attachment 2. process the email body to get some information from it 3. send an outbound email to an email address based on the information derived from step 2. The email will include the attachment received in step 1 I'm not sure where to start with this one so any suggestions would be appreciated.

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=UTF-8" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000066"> <font size="+1">Hi all,<br> <br> I am interested in

> >>I'm not sure that any solution with the MySQL dialplan command is going to >>be ideal. You also can't nest your queries, ie the connectid/result id >>seems to only be >good for one resultset at a time... try doing something >>like findme/followme with that! > > Thanks > > What is a better way to do it then in terms of performance, security,

I am unable how to get a zomebie sip channel to hangup. I've tried the following in the manager but it doesn't work. Action: Status Response: Success Message: Channel status will follow Event: Status Privilege: Call Channel: SIP/2003-09e2bbe8<ZOMBIE> CallerID: 093611168 CallerIDName: <unknown> Account: State: Up Link: SIP/2003-09e719f0 Uniqueid: 1171346560.592 Event:

I wish to set up two PSTN calls and then connect them similar to Jajah (is this called 3pcc?). The PSTN interconnect is handled by a third party SIP provider. I can do this using the manager or call files. An example (using php) would be: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n");