Displaying 20 results from an estimated 7000 matches similar to: "Asterisk 12 Outbound Authentication Failures on Realm"
2015 Mar 15
2
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic
configuration works, and I am connected to a SIP trunk using SIP.US, and
have set up my inbound calling which works correctly (when I call my PBX
DID, the call does come into my PBX network).
The issue is that I am not able to make outbound calls, because the call
fails with the error:
2015 Mar 15
3
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
That was the issue, thanks. I now am able to get the caller ringing on an
outbound call, but an external phone number (E164) I am dialing does not
ring.
On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com
> wrote:
>
>
> On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan <
> sonny.rajagopalan at gmail.com> wrote:
>
>> I have setup my
2015 Mar 15
4
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
Yes, I think the dial does get executed (sonny calling outbound
202-555-1212):
core set verbose 3
Console verbose was OFF and is now 3.
-- Executing [912025551212 at from-internal:1] Log("PJSIP/sonny-00000031",
"NOTICE, Dialing out from "" <sonny> to 12025551212 through fromgw") in new
stack
[Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @
2015 Mar 15
0
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> That was the issue, thanks. I now am able to get the caller ringing on an
> outbound call, but an external phone number (E164) I am dialing does not
> ring.
>
Any error messages? If you set 'core set verbose 3' and try it, does the
Dial get executed?
>
> On Sun, Mar
2015 Mar 15
0
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> Yes, I think the dial does get executed (sonny calling outbound
> 202-555-1212):
>
> core set verbose 3
> Console verbose was OFF and is now 3.
> -- Executing [912025551212 at from-internal:1]
> Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from
2015 Mar 15
0
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic
> configuration works, and I am connected to a SIP trunk using SIP.US, and
> have set up my inbound calling which works correctly (when I call my PBX
> DID, the call does come into my PBX network).
>
> The
2018 Apr 16
2
PJSIP error No auth credentials for realm(s) 'asterisk' in challenge
Hi all,
we are trying to move our servers from chan_sip to chan_pjsip. At this
time no problems with phones, they all register fine and can place
calls. But for a trunk we face problem and can't place calls despite the
fact that registration is OK. What we get is:
[2018-04-16 16:08:33] WARNING[18665]:
res_pjsip_outbound_authenticator_digest.c:178
2016 Aug 15
2
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
Hello
using pjproject 2.5.5
using asterisk-certified-13.8-cert1
Compiled pjproject 2.5.5 with :
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr
--libdir=/usr/lib64 --enable-shared --disable-video --disable-sound
--disable-opencore-amr
Compiled Asterisk 13 with
./configure --libdir=/usr/lib64
All pjproject modules are selectable in menuselect, so here no problem.
2012 Oct 09
2
Asterisk sends wrong fxs 'Idle' hints
Hi,
I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a
remote peer and an fxs phone gets connected and the remote peer hangsup,
then asterisk sends the "Idle" state to notify the watcher before you
hangup the fxs phone! Such a way if the user forgets to hangup the fxs
phone (which is a cordless for example) then the operators will keep
sending calls to him
2006 Jan 17
9
using "find" when you have 2 has_many relations
An Account has_many Websites which in turn has_many WebsiteDomains
Now I can of course do this:
@domains = Account.find(1).websites.find(1).website_domains.find(:all)
To get all the domains for Account with id 1 and Website with id 1.
I would like to do something like this though:
@domains =
Account.find(1).websites.find(:all).website_domains.find(:all)
IE, get ALL domains for all websites in
2006 Feb 13
2
[PATCH] Allow generic autocompleter (Ajax.Watcher)
Hey. Below is a patch to allow generic Ajax.Autocompleters. Basically
it''s for people who wanna be able to watch an input for changes, but
don''t want it to pop up an autocompleter box below. Useful for live
previews, that kind of thing.
Someone''s gonna have to fix the tabstops. I couldn''t be arse to work out
how to make vim do the softtabs properly. Spaces for
2006 Nov 29
7
how to debug context switching and mutex contentions?
I''m looking for a suggestion on a good way to hunt down the source of
high context switching and mutex contentions...
Is dtrace the way to go now, or should I stick with something like lockstat?
Russ
This is a 5 second interval for mpstat:
CPU minf mjf xcal intr ithr csw icsw migr smtx srw syscl usr sys wt idl
16 0 0 1115 1241 206 9095 912 2420 7393 0 12105 68 25
2020 Jan 19
2
Asterisk16 - PJSIP - Error 401 on outbound registration
Le 19/01/2020 à 00:31, Joshua C. Colp a écrit :
> On Sat, Jan 18, 2020 at 1:14 PM Administrator <admin at tootai.net
> <mailto:admin at tootai.net>> wrote:
>
>
> Le 17/01/2020 à 11:54, Administrator a écrit :
> >
> > Le 15/01/2020 à 19:24, Administrator a écrit :
> >> Hi all,
> >>
> >> we face a strange
2010 Mar 04
4
Permissions problem
What am I doing wrong here? I need to be able to write to /var/cvs.
This used to work before I moved these groups into an LDAP directory
instead of /etc/group:
[scarolan at watcher:/var/cvs]$ touch test.txt
touch: cannot touch `test.txt': Permission denied
[scarolan at watcher:/var/cvs]$ ls -ld
drwxrwsr-x 4 cvs cvsgrp 4096 May 18 2008 .
[scarolan at watcher:/var/cvs]$ id scarolan
2011 Apr 16
4
Jabber / GTalk / hints
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi!
Are hints not yet implemented in res_jabber?
I have this here:
exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com
But the hint doesn't show any difference. It always shows online on the
phone and core show hints always shows that:
6003 at internal : SCCP/6003 State:Unavailable Watchers 0
6002 at internal :
2015 Jan 19
1
Meaning of core show hint output
Hi all
If I have the following in my dialplan:
exten=>25001,hint,SIP/25001
Doing a
core show hint 25001
results in
25001 at local : SIP/25001
State:Idle Watchers 0
1 hint matching extension 25001
in the Asterisk CLI.
What does the
Watchers 0
mean?
I use the hints table output via core show hints for logic in my dialler
application - but
2016 Apr 01
5
Asterisk 13.8.0 alembic database update fails.
On Fri, Apr 1, 2016 at 3:22 PM, George Joseph <george.joseph at fairview5.com>
wrote:
>
>
> On Fri, Apr 1, 2016 at 3:15 PM, Harley Peters <harley at thepetersclan.com>
> wrote:
>
>> On 04/01/2016 04:06 PM, Joshua Colp wrote:
>>
>>> Harley Peters wrote:
>>>
>>>> I get the following error when trying to update date the database
2020 Jan 18
2
Asterisk16 - PJSIP - Error 401 on outbound registration
Le 17/01/2020 à 11:54, Administrator a écrit :
>
> Le 15/01/2020 à 19:24, Administrator a écrit :
>> Hi all,
>>
>> we face a strange behavior while connecting an Asterisk16 instance
>> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of
>> them having Kamailio as front-end. With other providers -we don't
>> know if they run
2015 Oct 18
3
pjsip show xxxx like endpoint?
Did you open a Jira issue for this yet? I can actually work on this this
week.
On Fri, Oct 16, 2015 at 9:44 AM, George Joseph <george.joseph at fairview5.com>
wrote:
> On Fri, Oct 16, 2015 at 4:00 AM, Bryant Zimmerman <BryantZ at zktech.com>
> wrote:
>
>> Is there a way to limit the items returned by pjsip show [type] using like
>>
>
> There isn't but
2006 Feb 24
1
Polycom IP 601 Buddy Watch doesn't work after Asterisk reload
Hi,
I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the "show hints" command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured.
Before the reload in the CLI appears:
-= Registered Asterisk Dial Plan Hints =-
3002 : SIP/3002 State: