Displaying 20 results from an estimated 3000 matches similar to: "PJSIP error No auth credentials for realm(s) 'asterisk' in challenge"
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 04
1
PJSIP: Failed to create outgoing session to endpoint
Hello.
I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip.
Now i am transfering all from chan_sip to chan_pjsip. And have a lot of
questions. First of...
system: Asterisk 13.2 on slackware 14.1
Errors on outgoing call:
[2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778 request: Failed to
create outgoing session to endpoint 'srv_d228'
[2015-03-03 00:18:58]
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 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
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
2020 Apr 06
2
Outgoing PJSIP using Kamailio
Hello,
We have a provider which is using Kamailio as front end. Our asterisk
13/chan_sip server has no problem to register and pass/receive calls
form this provider.
Now we want to move to asterisk 16/pjsip and face problem. Registration
is OK but when we pass a call our INVITE never receive answer from the
provider. We opened a ticket to their support but in the mean time we
want to know
2015 Mar 11
2
PJSIP some AMI events is absent?
Hello.
Asterisk 13.2, PJSIP.
Problem: I do not get any AMI events when changing the status of the
contact.
When using chan_sip I got "peerstatus" event.
When using res_pjsip and devices (endpoint configuration) I got
"peerstatus" event.
When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND
AUTHENTICATION i got "registry" event.
When using
2023 Jun 08
1
Problem with pjsip
Hello everyone.
I allow myself to submit a problem that I can not solve with my VOIP
provider Orange in France
[2023-06-08 13:19:03] ERROR[185091]:
res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error
configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid
character '@'
[2023-06-08 13:19:03] ERROR[185091]: config_options.c:798
aco_process_var:
2019 Jan 26
3
INVITE from DID: No matching endpoint found but completes the call anyway
I have a trunk set up for the DID from my provider:
[my_provider]
type=registration
outbound_auth=my_provider
server_uri=sip:sip.example.com
client_uri=sip:my_username at sip.example.com
retry_interval=60
[my_provider]
type=auth
auth_type=userpass
password=123456
username=my_username
[my_provider]
type=aor
contact=sip:sip.example.com:5060
[my_provider]
type=endpoint
context=from-my_provider
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.
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.
George,
I have the detailed log below. (Resending after trimming the email to 40KB.)
The sequence below just repeats ad-nauseam. Is this a SIP trunk issue?
Thanks!
---------------------
Transmitting SIP request (885 bytes) to UDP:65.254.44.194:5060 --->
INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0
Via: SIP/2.0/UDP 18.18.19.123:5060
2017 Sep 26
2
asterisk pjsip as voip client with multiple registrations
hi,
i want use asterisk+pjsip as voip client with multiple registrations
(perf testing)
i'm using this example configuration for one account
[308]
type=registration
outbound_auth=308
server_uri=sip:308 at example.com:5060
client_uri=sip:308 at example.com:5060
[308](auth-userpass)
username=308
password=pass
[308](aor-single-reg)
contact=sip:example.com:5060
[308](endpoint-basic)
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
Hi all,
I have an issue with asterisk 13 and pjsip. I guess it is somehow
Firewall related, but I'm unsure.
A registration to Sipgate is established successfully:
<Registration/ServerURI..............................> <Auth..........>
<Status.......>
==========================================================================================
2013 Sep 03
1
Asterisk 12 Outbound Authentication Failures on Realm
Trying out Asterisk 12 and the new pjsip module...
When I create an registration object that links to an auth object, the
registration fails with "res_pjsip_outbound_authenticator_digest.c:90
digest_create_request_with_auth: Failed to create new request with
authentication credentials" unless the auth object has it's realm set
exactly to the realm returned in the 401 response from
2015 Mar 16
1
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 5:56 PM, Sonny Rajagopalan
<sonny.rajagopalan at gmail.com> wrote:
> George,
>
> I have the detailed log below. (Resending after trimming the email to 40KB.)
>
> The sequence below just repeats ad-nauseam. Is this a SIP trunk issue?
>
> Thanks!
>
I don't see anything obvious. The registration works though, right?
You might want to compare
2014 Jul 16
1
PJSIP outbound register and inbound calls
Hi all,
In my case I using realtime,
here is how it looks in plant
[10001]
type=registration
transport=upd_static
outbound_auth=10001
server_uri=sip:600 at 192.168.1.1:5060
client_uri=sip:600 at 192.168.1.4:5060
[10001]
type=auth
auth_type=userpass
password=600
username=600
[10001]
type=aor
contact=sip:192.168.1.4:5060
[10001]
type=endpoint
transport=upd_static
context=dialmap
disallow=all
2023 Jun 21
1
PJSIP not performing outbound authentication
Dis you set "outbound_auth" in your endpoint configuration to Twilio?
On 21/06/23 11:19, TTT wrote:
> I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP
> (Twilio) who requires outbound authentication. My pjsip.auth.conf contains:
>
> [Twilio]
> type=auth
> auth_type=userpass
> password=mysecret
> username=myun
>
> However, my calls
2015 May 21
4
PJSIP CCSS
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Hash: SHA1
Le 21/05/2015 00:16, Joshua Colp a ?crit :
> If CCSS is needed then the only option is to use chan_sip. The
> chan_pjsip module does not implement CCSS in any way.
Is CCSS support planned for PJSIP? chan_sip is in "extended" state in
asterisk-13, so chan_pjsip should be preferred for new installations, ri
ght?
Thanks,
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