11.0: How To Get Remote Commands To Show On Cli?


I'd like to see on cli what happens on executing remote commands. For instance:

asterisk -rx "originate Motif/gvoice/12026668765@voice.google.com,,rL(5000)) extension s@default"

Now I get on cli, verbose 10:

-- Remote UNIX connection -- Remote UNIX connection disconnected

Any way to see call progress for remote commands? Something is going wrong with this command, although it works fine from the dialplan.


Asterisk Users 2.9 years ago 0 Answers

Confbridge Command Not Found


Currently running version and trying to manage confbridge rooms and users. When I try to use the confbridge cli command I get a command not found error.

CLI> confbridge No such command 'confbridge' (type 'core show help confbridge' for other possible commands)

I've tried googling this but did not get anywhere. How can I enable the confbridge commands?


Asterisk Users 3.1 years ago 1 Answer

Question About Cli


Hello guys, i would like to ask a question about cli.

Today, while i was using the cli, i thinked that there could be more features. IMHO, might be interesting, for example, to add a sip extensions from cli, or other similar functions, without having to modify the configuration files.

Or not? What do you think?


Asterisk Users 3.1 years ago 2 Answers

Graceful Restart



Is there a way to detect, via cli or any other way, that Asterisk is in "graceful shutdown" mode, not accepting any new calls? Or to put the question a different way, how can I know that Asterisk has restarted again after the command "core restart graceful" in an automated way?

Best regards, Jan Blom

Asterisk Users 3.2 years ago 3 Answers

IAX2 Registered OK without IP


This has come up before on the list and archives but I don't seem to
find a solution for this. On just a few nodes we have this situation
where we see the IP disappear from the CLI iax2 show peers list but
the status shows OK: 3012/3012 (Unspecified) (D) 0 OK (89 ms) How can the status be OK a few milliseconds ago and have no IP ?? The
strange thing is that the IP does show up once in a while and then
disappears once again but the OK is always there.…

Asterisk Users 3.4 years ago 4 Answers

Call Forwarding


Hi Guys, Seeing an issue with and also When we do call forwarding if the call coming in to be forwarded
asterisk sends the invite out to our ITSP as
username@anonymous.invalid instead of username@domain. When call comes in with CLI and is forwarded it sends it as
username@domain to our ITSP. Is this a bug or is there something I need to turn on or off? All the
ITSP's we use authenticate on username and domain. Thanks

Asterisk Users 3.4 years ago 0 Answers

add new sip account in sip.conf with API Action UpdateConfig with php


Hi List, I am trying to add new SIP account in new file additional_sip.conf. I read
in Wiki there is API command UpdateConfig which is used to update , add and
delete any entry from configure files. I am using PHP to make new entry in
additional_sip.conf. Below is the code which I tryed .... < ?php
$socket = fsockopen("","5038", $errno, $errstr, 30);
if (!$socket)
} else {
fputs($socket, "Action: Loginrn");
fputs($socket, "UserName: adminrn");
fputs($socket, "Secret: adminrnrn"); fputs($socket, "Action: UpdateConfigrn");
fputs($socket, "reload=yesrn");
fputs($socket, "SrcFilename: additional_sip.confrn");

Asterisk Users 3.4 years ago 1 Answer

Operation not permitted ??


Anyknow know this problems ?
I read on the net that it's a possible network problems, but i don't think
because it's a VMWare server and in the same server i have other
asterisk without this problems. best regards
Le 25 avril 2012 09:35, Olivier CALVANO a écrit :
> Hi
> i have a lot of error in the CLI of one of my Asterisk:
> [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit
> of 0x8ef2130 (len 867) to returned…

Asterisk Users 3.5 years ago 2 Answers

FXO -> GSM Gateway Problem


Thanks Dhaval for taking the time to look at my question. I have tried to print the hangup cause however as you can see below it
doesn't show that section of the dialplan. I have ammended below the CLI and extensions.conf with the changes I made. ASTERISK CLI == Using SIP RTP CoS mark 5

Asterisk Users 3.5 years ago 0 Answers

404 Response to Invite - Should be 401


Hi all, This post is in case someone else has this problem. The cause of the issue turned out to be one of the site technicians
having the same extension registering from his laptop as the ATA we
were testing. His laptop wasn't always connected to the voice network
and the soft-phone wasn't always on. Sometimes we were able to make
calls from the ATA we were testing and sometimes we would get the
problem described below. Everything came to light when his soft-phone registered (throwing up a
non-voice network IP address) whilst I…

Asterisk Users 3.5 years ago 0 Answers