I have an ARI application that is registered for Stasis in the dialplan.
One of the events I reap in my application is a ChannelDtmfReceived. The
thing is, Asterisk 13.6.0 sends me two DTMF for each DTMF pressed (have
tried both SIP phones and landlines). That is, I receive two
ChannelDtmfReceived
e Joseph <george.jos...@fairview5.com
> wrote:
>
>
> On Thu, Feb 18, 2016 at 8:20 PM, Sonny Rajagopalan <
> sonny.rajagopa...@gmail.com> wrote:
>
>> Thanks George, for your mighty quick response.
>>
>> I made the changes (re: server_uri_pattern etc.) and still, no
Feb 18, 2016 at 9:56 PM, George Joseph <george.jos...@fairview5.com>
wrote:
>
>
> On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan <
> sonny.rajagopa...@gmail.com> wrote:
>
>> Hello,
>>
>> I have an Asterisk 13.6.0 PBX using PJSIP connected to the
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
.
Hope this helps.
Thanks again!
On Wed, Feb 17, 2016 at 3:48 PM, George Joseph <george.jos...@fairview5.com>
wrote:
>
>
> On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan <
> sonny.rajagopa...@gmail.com> wrote:
>
>> Wow. Incredible. That worked. The bac
:43 PM, George Joseph <george.jos...@fairview5.com
> wrote:
>
>
> On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan <
> sonny.rajagopa...@gmail.com> wrote:
>
>> I made some progress. The first thing I have realized is that it is my
>> Twilio configuration i
n, the same asterisk configuration on the same machine connected to
the same twilio SIP trunk worked for UDP)
If anyone knows the trick to make pjsip_wizard.conf work with twilio, I
would very much appreciate any insight...
Thanks,
Sonny.
On Wed, Feb 17, 2016 at 8:38 AM, Sonny Rajagopalan <
sonny.rajag
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_l...@earthshod.co.uk>
wrote:
> On Wednesday 17 Feb 2016, Sonny Rajagopa
nly this line was changed.
On Wed, Feb 17, 2016 at 8:28 AM, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:
> OK. I will report with my findings. It appears increasingly likely that I
> have done something very silly on my side. It is a little perplexing that
> the EXACT setup (o
OK. I will report with my findings. It appears increasingly likely that I
have done something very silly on my side. It is a little perplexing that
the EXACT setup (on the same machine) worked for UDP ...
On Wed, Feb 17, 2016 at 8:23 AM, Joshua Colp <jc...@digium.com> wrote:
> Sonny Ra
perplexed as to
why Asterisk wouldn't consume those TCP segments; the port is owned by
Asterisk.
On Wed, Feb 17, 2016 at 8:15 AM, Joshua Colp <jc...@digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>> Is there a specific place where I can set logger to log incoming TCP
&
:* LISTEN
10313/asterisk
tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN
10313/asterisk
On Wed, Feb 17, 2016 at 7:57 AM, Joshua Colp <jc...@digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>> I receive a TCP ack back from that port (5060;
tried a fresh build with protocol=tcp.
Did not work.
On Wed, Feb 17, 2016 at 6:35 AM, Joshua Colp <jc...@digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>> I can confirm that the server is receiving the SIP request, but simply
>> doesn't do anything with it (log from
ntent-Length: 0
On Mon, Feb 15, 2016 at 6:01 PM, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:
> 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 <jc...@digiu
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 <jc...@digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>> Does this help:
>>
>
> Yes, the transport parameter is
Length: 0
On Mon, Feb 15, 2016 at 2:53 PM, Joshua Colp <jc...@digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>> Thanks for the mighty quick response, Joshua!
>>
>> I am using Zoiper on Linux softclient:
>> REGISTER sip:;transport=TCP SIP/2.0
>>
>
Thanks for the mighty quick response, Joshua!
I am using Zoiper on Linux softclient:
REGISTER sip:;transport=TCP SIP/2.0
Changed the port back to 5060.
On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jc...@digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>
>
>
>
This question was asked by Chirag on March 4 2015 earlier, but I am
following exactly the same procedure here and I cannot even get my clients
to register on Asterisk.
Here's my PJSIP.conf:
[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5061
...
[endpoint_internal](!)
type=endpoint
alplan from a
>> script and force a reload.
>>
>> To directly answer you question I do not believe there is an API baked
>> into asterisk to update the pjsip.conf and extensions.conf directly from
>> the dialplan.
>>
>> Thanks
>>
>> Bryant
>>
&g
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.
--
_
for someone who stumbles into this issue...
On Sat, Dec 12, 2015 at 6:05 PM, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:
> Hello,
>
> I am developing a voicemail application and was wondering if I can
> redirect a caller to voicemail using ARI. Any help is appreciated.
&
Hello,
I am developing a voicemail application and was wondering if I can redirect
a caller to voicemail using ARI. Any help is appreciated.
Thanks,
Sonny.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Thanks Annus, Amit.
Yes, Amit, the plus sign in front is necessary. I was able to get this to
work by changing the codecs that the SIP trunk will use. I had to set up
wireshark on my Asterisk instance, gather that it didn't work for a 488 SIP
"No acceptable here" message, which led me to
no point do I point it to the
from-internal context.
On Wed, Dec 2, 2015 at 10:35 AM, Annus Fictus <annusfic...@gmail.com> wrote:
> Maybe is because now it's a different context:
>
> from-twilio-remove-plus
>
> before
>
> from-internal
>
> is right?
>
> regard
Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING,
TWILIO)). It does not work and NO error message in CLI.
I have also tried
http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I
first emailed this group, but that does not seem to work either.
Here is my
Hello,
I am running Asterisk 13.6.0 in an AWS instance, and I set it up with
Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the
calls actually "reach" the PBX, but for some reason, they are not caught by
any of my extensions context.
Here's what I observe when I test this
I figured this out. The issue was that I did not have a
type=global
configuration in the [global] context. It should be:
[global]
type=global
default_outbound_endpoint=SillyEndpoint
...
[SillyEndpoint]
type=endpoint
etc.
On Sun, Nov 29, 2015 at 2:33 PM, Sonny Rajagopalan <
sonny.rajag
I am trying to set up a default outbound endpoint for my Asterisk 13.6.0
PBX, and per
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip,
I do in pjsip.conf:
[global]
default_outbound_endpoint=SillyEndpoint
...
[SillyEndpoint]
type=endpoint
etc.
However, when I check
(MESSAGE(body)="[${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}]
> Your message to ${EXTEN} has failed. Retry later.")
> exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
> exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
> exten => _.,n,MessageSend(${ACTUALFROM},Ser
hen I create that
> message_context in the extension.conf, and it works.
>
> On Tue, Nov 17, 2015 at 9:34 AM, Sonny Rajagopalan <
> sonny.rajagopa...@gmail.com> wrote:
>
>> So, the only thing that is needed in the endpoint definition in
>> pjsip.conf (there is no such file pjsip.endpoint_
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,
<100>
>
> dtmf_mode=rfc4733
>
> use_avpf=no
>
> ice_support=no
>
> media_use_received_transport=no
>
> trust_id_inbound=yes
>
> media_encryption=no
>
> rtp_symmetric=yes
>
> rewrite_contact=yes
>
> *message_context=astsms*
>
>
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
anyone successfully done SIP trunk registration with PJSIP in Asterisk
13.1.0?
On Sun, Mar 15, 2015 at 12:34 PM, Sonny Rajagopalan
sonny.rajagopa...@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
Hello,
I am trying to force a registration and unregistration with my SIP trunks,
but I see pjsip send unregister, but no register.
I.e., I am looking for pjsip send register.
Is there any such command? If so, why do I not see it in my CLI?
Should I upgrade to 13.2.0?
Any insight appreciated.
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.jos...@fairview5.com
wrote:
On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan
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
-555-1212 does not ring.
at hangup on caller (sonny):
== Spawn extension (from-internal, 912025551212, 2) exited non-zero on
'PJSIP/sonny-0031'
On Sun, Mar 15, 2015 at 3:25 PM, George Joseph george.jos...@fairview5.com
wrote:
On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan
is not happy with this.
Thank you for responding!
On Sunday, March 15, 2015, George Joseph george.jos...@fairview5.com
wrote:
On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan
sonny.rajagopa...@gmail.com
javascript:_e(%7B%7D,'cvml','sonny.rajagopa...@gmail.com'); wrote:
Yes, I think the dial
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
Yes, the registration works. I'll check the auth on registration.
On Sun, Mar 15, 2015 at 9:37 PM, George Joseph george.jos...@fairview5.com
wrote:
On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan
sonny.rajagopa...@gmail.com wrote:
George,
I have the detailed log below. (Resending
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
wrote:
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
Sorry, but I should have RTFMed my follow up question. I used
${CALLERID(num)} to get it.
Thanks.
On Friday, March 13, 2015, Sonny Rajagopalan sonny.rajagopa...@gmail.com
wrote:
Oh, wow! Changed it and now I am getting calls into my context (fromgw).
Unfortunately, the actual caller ID
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
external_media_address=publicIPOfEC2Instance
external_signaling_address=publicIPOfEC2Instance
;Templates for the necessary config sections
[endpoint_internal](!)
type=endpoint
context=from-internal
disallow=all
allow=!all,ulaw
direct_media=no
rtp_symmetric=yes
On Thu, Mar 5, 2015 at 5:52 PM, Sonny
have helped me tremendously.
On Thu, Jan 8, 2015 at 8:03 PM, Sonny Rajagopalan
sonny.rajagopa...@gmail.com wrote:
Well, I thought it worked, but it actually doesn't--I am able to get the
caller pick up the phone, but for some reason, I cannot hear anything on
either side no matter who does
|
|(VM) |
| 192.168.1.239 |
|---|
On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan
sonny.rajagopa...@gmail.com wrote:
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
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
that it is retaining state
for).
On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan
sonny.rajagopa...@gmail.com wrote:
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
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