Displaying 20 results from an estimated 1000 matches similar to: "How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?"
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,
2015 Sep 22
2
How to set the global setting for each pjsip endpoint
I have many endpoints and each endpoint has some parameter in common so i
wonder is there any way to config one for all endpoints? Like in my example
I have two endpoints and I repeat the same thing,
[100]
type=endpoint
aors=100
auth=100-auth
allow=ulaw,alaw,gsm,g726
context=from-internal
callerid=device <100>
dtmf_mode=rfc4733
use_avpf=no
ice_support=no
2015 Sep 22
2
How to set the global setting for each pjsip endpoint
how if I use the auto generate once from freepbx ?
On Tue, Sep 22, 2015 at 10:12 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote:
>
>
> On 22 September 2015 at 16:04, Thyda ENG <engthyda at gmail.com> wrote:
>
>> I have many endpoints and each endpoint has some parameter in common so i
>> wonder is there any way to config one for all endpoints? Like in my
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 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
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 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 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
2015 Sep 22
2
How to config instance messaging for asterisk 12
I am using the asterisk 12 with pjsip, I wonder how could I config the
instance meesseging for pjsqip in asterisk 12 ? What is the default message
context for pjssip ? I use the default extension.conf from the installation
and I successfully could make the call over each but when I try to send
message, it does not receive by the client.
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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
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 Jan 08
2
Asterisk 13.1.0/PJSIP peer IP address issue
I am following the instructions in
https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am
trying to make a call from extension Alice (6001) to extension for Bob
(6002). When I make the call, I can hear the ringing on Alice's phone
(caller), but Bob's phone (callee) doesn't ring, or show a call coming in
from Alice. My setup and environment is as follows: Alice, Bob
2016 Jul 01
2
CALLERID on pjsip doesn't work?
Asterisk 13.8
Is CALLERID(all) supposed to wok for pjsip? When I do this:
exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
same => n,Dial(PJSIP/phone123, 30)
I expect the callerid to be as set, but is always seems to be "phone123",
the name of the endpoint.
Andrew
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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
2016 Jan 29
3
Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API
Hi,
I am using Asterisk 13.6.0 and was wondering if I can programmatically add
users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk
server using API of some sort.
Please do let me know.
Thanks,
Sonny.
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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
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 15
2
Asterisk 13.6.0/The simplest TCP configuration does not work
Nope, there are no contacts to show that pertain to these endpoints (only
my SIP trunks show up).
On Mon, Feb 15, 2016 at 5:31 PM, Joshua Colp <jcolp at digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>> Does this help:
>>
>
> Yes, the transport parameter is in the Contact header so it's interesting
> it didn't work. If you use pjsip show contacts what