similar to: Problems dialing to another Asterisk server

On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:07:35 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > <snip> > >> Here

On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:55:28 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > > > > > On

Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > Tested with X-Lite and it worked fiine. Is there some way to replace > "Anonymous" with a config parameter? > > Thanks for your kind help > > ---------------------------------------- > > From: murthy64 at hotmail.com > > To: asterisk-users at lists.digium.com >

Hello! Is it legal to use RedirectAction to redirect a call that is waiting in a queue? The idea is to have an external application manage a queue via manager API. The queue would merely collect calls and play moh. I've tryed this already but asterisk sends SIP/Forbidden to the channel in queue, after the channel has been redirected by RedirectAction, even though the response to

Hello! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and now want to add a SIP-Diversion header that changes the CallerID of a number that is not available on the trunk, the CallerID to be visible externally

Hello, We are moving from an older asterisk/SIP to a newer (18+) asterisk/PJSIP and trying to figure out how to add [identity] header when originating a call from AMI/PAMI. In the older version we would just set a variable like this: $action = new OriginateAction("SIP/...."); $action->setVariable('__SIPADDHEADER51',"Identity: $identity"); // $identity

Hi everyone, I want to call from one Asterisk to another Asterisk via SIP, but i dn't know how. I have found out something in these links: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels but I don't understand them very well. At first, I tried simply doing this: In SIP Client:

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

Hi. I'm making calls via the Manager OriginateAction. My action is set to be async and therefore I receive originiate events. Within the originate event that I receive there is a reason code. In the event of failure I need to dermine why the call failed (no pickup, rejected, no such number, circuit busy, ect) and inform the user with a meaningful message. I assume that one is suppose to

Hello Everybody, I am working on Fastagi and I am making use of Asterisk-java. But I don't find the class file for SIPPeersAction. Hence I am getting the error message when compiling my java code. ---------------------------------------------------------------------------- ------------------------------------------------ [root@localhost asterisk-java-0.1]#

hi, afaik, the action-id provided with the OriginateAction should only show up in the OriginateSuccess or OriginateFailure event. Intermediate events that are generated when the channels are create will NOT carry the action-id of the originate. The async flag tells asterisk to process originates in parallel, i.e. if you have two users originating calls and NO async flag set, the second originate

Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI> -- Executing Dial("SIP/16008-3d17", "SIP/16007&SIP/16006|20|tr") in new stack -- Called 16007 -- Called 16006

Hi, I have been running nut successfully for a long time with my Debian 11 server. I upgraded my server to Debian 12 today, which upgraded nut also from 2.7.4-13 to 2.8.0-7. I noticed that after upgrade there was a upssched process running and taking 100% cpu time. I checked if there were any changes to configuration file formats with nut upgrade and only differences I noticed were a terminology

Hello, We have a webrick server running our nice app, and an apache server being used to serve the rest of the site and act as a proxy for the webrick app. <code> <IfModule mod_proxy.c> ProxyRequests Off <Proxy *> Order deny,allow Allow from all </Proxy> ProxyPass /appname http://server.com:3000 ProxyPassReverse /appname

Dear all, as you know, we can use Originate Command to auto-dial a out-bound call to a extention or app since 1.6.2. but when i Originate a call, and hangup. the cdr of this call has no CDR(clid) and CDR(src). Could you tell me how to set the Callerid to cdr from an Originate call? I use Originate directly in the dialplan not AMI, so i can't set the callerid property like AMI use

Hello, I am using Asterisk-java, the Manager. And I have a problem I don't know how to sort it out!: Sometimes, when I send an OriginateAction my code receives an exception with this message: "Timeout waiting for response to Originate" I don't know what it means as Asterisk receives the action and then dials to the telephone, might anybody show me what is the problem???

Hi, I ran the strace command while upssched was 100% CPU hungry. This is what I got: 1686633611.702798 read(7, "", 1) = 0 <0.000004> 1686633611.702816 read(7, "", 1) = 0 <0.000004> 1686633611.702834 pselect6(11, [7 10], NULL, NULL, {tv_sec=1, tv_nsec=0}, NULL) = 1 (in [7], left {tv_sec=0, tv_nsec=999998800}) <0.000006> 1686633611.702862 read(7,

Hi, I ran the strace command while upssched was 100% CPU hungry. This is what I got: 1686633611.702798 read(7, "", 1) = 0 <0.000004> 1686633611.702816 read(7, "", 1) = 0 <0.000004> 1686633611.702834 pselect6(11, [7 10], NULL, NULL, {tv_sec=1, tv_nsec=0}, NULL) = 1 (in [7], left {tv_sec=0, tv_nsec=999998800}) <0.000006> 1686633611.702862 read(7,

With Asterisk-Java the proposed solution to connect to multiple Asterisk servers is to create multiple AsteriskManagerConnection obeject. Each ManagerConnection handles its own thread so there is no need for custom thread handing code. All you have to do is to make sure is the EventListener objects you pass to these connections synchronize access to shared data (if there are such accesses). I