Displaying 20 results from an estimated 900 matches similar to: "Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge."
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 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
2015 Mar 13
2
PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found
I have a working Asterisk 13.1.0 running, and I am trying to configure a
SIP trunk for outbound and inbound calling, and a DID for the Asterisk
server, which is used for incoming calls from PSTN.
I configured my SIP.US trunks (showing one gateway, gw1, here for brevity,
have two: gw1 & gw2, which are both configured on my end):
[sonnyGW1]
type=registration
transport=transport-udp
2015 Mar 25
0
PJSIP configuration for Asterisk 13.1.0/SIP trunk outbound calling
Hello,
I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0
and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the
appropriate ports. The SIP clients can be anywhere on the Internet,
including behind NATs.
I am able to get to my Asterisk server's internal extensions via the DID
(and appropriate dialplans) but I am not able to make outbound calls to
2015 Mar 13
1
PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found
Oh, wow! Changed it and now I am getting calls into my context (fromgw).
Unfortunately, the actual caller ID (6175551212) is not getting passed (but
I know Asterisk is getting this). How do I "reap" this actual caller ID in
my dialplan?
On Fri, Mar 13, 2015 at 4:55 PM, Joshua Colp <jcolp at digium.com> wrote:
> Sonny Rajagopalan wrote:
>
> <snip>
>
>
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
2015 Mar 13
0
PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found
Sonny Rajagopalan wrote:
<snip>
> [sonnyGW1]
> type=identity
> endpoint=sonnyGW1
> match=65.254.44.194
You want type=identify, not type=identity.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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
2015 Nov 17
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
Hello,
I am looking for documentation support for enabling instant messaging
between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as
Zoiper. Where do I enable this support on the server side and does it need
anything on the client side? I see plenty of online help for chan_sip, but
nothing for chan_pjsip.
I imagine there is both pjsip.conf configuration and extensions.conf
2015 Nov 17
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
So, the only thing that is needed in the endpoint definition in pjsip.conf
(there is no such file pjsip.endpoint_custom.conf) is
*message_context=astsms*
Is that correct? Anything I need to do in extensions.conf? I see that the
messages are received at Asterisk (when I turn on pjsip set logger on) but
they are not delivered to the other endpoint. What gives?
Any help appreciated. Thanks!
On
2015 Nov 17
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
Thanks again. How do you create that message context in extensions.conf?
On Mon, Nov 16, 2015 at 9:44 PM, Thyda ENG <engthyda at gmail.com> wrote:
> According to what I have done , I add the message_context to the
> pjsip.endpoint_custom.conf in /etc/asterisk and then I create that
> message_context in the extension.conf, and it works.
>
> On Tue, Nov 17, 2015 at 9:34 AM,
2016 Feb 17
2
Asterisk 13.6.0/The simplest TCP configuration does not work
Yes, it is enabled on port 5060. I do receive a TCP ACK back from the
server, so I know the TCP segment is received at the server hosting the
Asterisk build.
On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles <asterisk_list at earthshod.co.uk>
wrote:
> On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote:
> > OK. Let me ask this. Is anything else necessary, except choosing TCP as
> the
2015 Mar 05
2
PJSIP configuration for AWS/EC2 based Asterisk 13.1.0
Hello All,
I have an Asterisk server v13.1.0 running on EC2 and I am able to connect
and register SIP devices and "see" them on the asterisk CLI. I am also able
to place calls, but I am not able to hear any audio on either end after the
call is picked up.
I was wondering if you can tell me what a minimal configuration for
Asterisk on EC2 looks like. My current pjsip.conf configuration
2016 Feb 19
2
No matching endpoint found for incoming call from SIP trunk
Thanks George, for your mighty quick response.
I made the changes (re: server_uri_pattern etc.) and still, no luck--it
fails for the same error.
BTW, there is nothing for transport (but this is the same config from my
SIP/UDP + Twilio days, which worked):
*CLI> pjsip show transport twilio-siptrunk
Unable to find object twilio-siptrunk.
*CLI> pjsip show identifies
No objects found.
I did
2016 Feb 19
2
No matching endpoint found for incoming call from SIP trunk
Hello,
I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway.
I am able to make calls outbound through the gateway, but I am not able to
make calls into the PBX from external PSTN.
Specifically, an incoming call is _received_ by Asterisk, but it is not
able to route the call internally owing to the following error:
[Feb 18 21:08:47] NOTICE[4606]:
2016 Feb 17
2
Asterisk 13.6.0/The simplest TCP configuration does not work
On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> I made some progress. The first thing I have realized is that it is my
> Twilio configuration in pjsip_wizard.conf that was killing me. I have since
> removed that entire file from /etc/asterisk and I am able to make
> "from-internal" context calls (i.e., calls that do not
2015 Jan 08
4
Asterisk 13.1.0/PJSIP peer IP address issue
Thank you for your note, Scott.
I set rewrite_contact=yes for both contacts, and I also had to do
remove_existing=yes because I had to remove the existing contact
information (max_contacts = 1 was preventing new contact information)
using pjsip
qualify demo-alice etc., after which the right IP addresses showed in pjsip
show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is