similar to: Manger-command Getvar?

I found a clue as to why the second leg is not returning a local or remote address: [2023-07-06 11:40:35] WARNING[253072]: pjsip/dialplan_functions.c:903 channel_read_pjsip: No transport information for channel PJSIP/222-0000007d [2023-07-06 11:40:35] WARNING[935126]: func_channel.c:527 func_channel_read: Unknown or unavailable item requested: 'pjsip,local_addr' [2023-07-06 11:40:35]

On Tue, Jul 4, 2023 at 7:52 PM TTT <lists at telium.io> wrote: > Building on my last message, I am trying to get CHANNEL data using getvar > (through the AMI). And although I'm getting responses, some values > returned seem illogical. For example, phone 111 calls phone 222 via the > PBX. Here's the data I get back > > > > > > Channel A:

Building on my last message, I am trying to get CHANNEL data using getvar (through the AMI). And although I'm getting responses, some values returned seem illogical. For example, phone 111 calls phone 222 via the PBX. Here's the data I get back Channel A: "1688509741.112" , name: "PJSIP/111-00000064" , is originator: Y , call-Id: "u.l6kcou25cax60 at

Channel A: "1688509741.112" , name: "PJSIP/111-00000064" , is originator: Y , call-Id: "u.l6kcou25cax60 at mydomain.com <mailto:u.l6kcou25cax60 at mydomain.com> " , local_uri: "<sip:222 at mydomain.com <mailto:sip%3A222 at mydomain.com> ;user=phone>" , local_tag: "1734d973-c4da-4ae8-a37d-5f7065f1fe54" , local_addr:

Hia .... i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance

The following AMI command works well for all of the information I want: action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-00000028 Variable: CHANNEL(pjsip,XXXX) Where XXXX can be one of the many available items. However, when I create a call from A to B, all of the items return properly except: local_addr and remote_addr. More specifically, they return correctly for the A leg (that

Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read

Hi, package tiff-v3.5.7 contains the currently recommended version of libtiff in order to run spandsp (fax support for asterisk). Imho tiff-v3.5.7 is not very easy to find in the internet, and maybe will almost disappear, because it is an "old" version, I put it on our little asterisk download page. Maybe it'll help someone. It works fine together with the other asterisk stuff

Hi, some days ago we discused here the need for an analyzer for the 1000 Hz tone, as opposite application to Milliwatt. Here it is: Mwanalyze http://planinternet.net/download/voip/asterisk/app_mwanalyze.c It performs a Fourier analysis for a fixed frequency and tells the amplitude. The frequency is not limited to 1000 Hz, but can be passed as argument. The periode duration must be a mulitple

Hi, how can I disable event_log in order to reduce hard disk activity? I can't find any hints in conf files. Must I hack the source code or even use brutal methods like creating a dir called event_log in the log dir, in order to prevent asterisk from creating an event_log file? (Just chmod a-w event_log does not work, unfortunately.) Thanks for any hints! Roger.

Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's "einen Ruf heranholen". It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to

Hi, I have unpleasent short audio gaps when a perl based agi scripts starts. Thus, I now started to put all those things in C programmed daemons for fast-agi. Anyway I'm looking for another mean, which would help me more quickly. I noticed, that all agi scripts are running with system priority -11, like asterisk does. This is really waste of priority. I would like to have the AGI scripts

Hi, whatever dialplan I'm using for outgoing calls via PRI (Digium card, chan_zap), the callerid when receiving calls has no leading zeros, which are normally used to distinguish local, national and international calls in Europe. The number has always the area code in front, but the country code only for foreign calls. Now I'm looking for any mean to decide, whether the received

Remote Call Pick up feature is very much implemented in asterisk. You can pick up a call for another extension by dialing *8# To be able to do that, you need to have the extensions in the same pickup group, configurable through sip.conf and zapata.conf. -- sudhir > ------------------------------ > > Message: 14 > Date: Fri, 27 Aug 2004 14:17:26 +0000 (UTC) > From:

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hi, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes through the asterisk box in the middle. Using sip debug I even can't find

Hi, I wish to know if GNUGk can work with * running as a gateway with the Digium FXO card. Kindly share your experiences in case there are some issues which one must know before going in for such a setup. Also, I've been reading about the DialTone detection capability by the hardware in different countries. What are the issues with it? Thanks & Regards, Deepak ----- Original Message

Is there a way to tell the difference in an agi between the person actually hanging up the phone and the manager interface doing a hangup command? Thanks, Jerry

Hello Users......... I'm Now doing on Asterisk Manager for My knowledge Growth, Can anybody explan me on Asterisk Manager settings....... in manager.conf [general] enabled =yes port = 5038 bindaddr = 192.168.2.75 displayconnects = yes [hyperion] secret = hyperion permit=192.168.2.76/255.255.255.0 deny=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user write =

Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to