Displaying 20 results from an estimated 1000 matches similar to: "Issues with Twilio number incoming call and context matching"
2015 Dec 02
2
Issues with Twilio number incoming call and context matching
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 log:
[Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call
from
2010 Jul 30
1
VUC Friday: Twilio OpenVBX
Interesting offering, free from Twilio, this is php you install on
your own server to build a brandable "VBX". Worth checking out!
Listen to tomorrow for more about this and talk to lead engineer or
Twilio CEO if you have any questions;
sip:200901 at login.zipdx.com or Skype:vuc.me
IRC: #vuc on Freenode.net or http://vuc.me/irc
Info about VUC is htp://vuc.me
Best,
/r
2009 Dec 31
2
Twilio
http://www.techcrunch.com/2009/12/30/twilio-raises-3-7-million-for-power
ful-telephony-api/
wow really?
Cheers,
Dean
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2014 Feb 27
0
How to Integrate Twilio With Your Rails 4 App
*Disclosure: I am a Developer Evangelist at Twilio.*
Hey everyone,
I just published a blog post on how to use Webhooks and Concerns to
integrate Twilio into a Rails 4 application. I hope some of you might find
this useful:
https://www.twilio.com/blog/2014/02/twilio-on-rails-integrating-twilio-with-your-rails-4-app.html
I've also published the full source for this tutorial on Github:
2018 Feb 08
3
pjsip trunking configuration issue
Greetings !
My goal is to get Twilio trunking working, and with TLS/SRTP.
I see this concerning message in my log:
[Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ?twilio' from configuration file ?pjsip.conf?
Thus, ?pjsip show endpoints? does not show the endpoint for the Twilio trunk.
Hoping for a sanity check of
2023 Jun 21
1
PJSIP not performing outbound authentication
On Wed, Jun 21, 2023 at 05:19:11PM +0000, 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 using the trunk are rejected with a 403. Using pjsip
>
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
2023 Jun 21
1
PJSIP not performing outbound authentication
I didn't use pjsip_wizard, I'm kind of crafting this by hand as I learn. I actually have a plain asterisk, and a FreePBX, system to help me learn. I sometimes create something in FreePBX to see what it does to the config files. So that's how I modelled my pjsip.X.conf files
If I issue the command "pjsip show endpoint Twilio" it does show that outbound_auth=Twilio
Does
2023 Jun 21
2
PJSIP not performing outbound authentication
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 using the trunk are rejected with a 403. Using pjsip
logging I notice that the outgoing invite does not have an authentication
line. Why is Asterisk not sending
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
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
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]:
2019 Mar 29
3
why doesn't extension "s" work ?
I'm using "s" extension in my dialplan:
[gv-voice]
exten => s,1,Verbose(callerid is "${CALLERID(all)}" or
"${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To)
same=>n,....
But when a call comes in to the gv-voice context, "s" doesn't match the
extension:
res_pjsip_session.c:2991 new_invite: Call from
2023 Jul 01
1
SetCallerPres command gone
The AGI debug command worked well, and I found the offending command:
SetCallerPres(allowed)
That worked in Asterisk 13, but from my google searching it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13). I thought it was replaced with CALLERPRES(allowed) but this generated an error too in Asterisk 20.
Is there a replacement command?
-----Original
2023 May 23
3
Problems Solved, two left
And I think they're both small.
Solved: tcpdump showed no packets coming in, so I went to my DID
provider's Website to discover to my intense embarrassment that the DID
number had been set up forwarded to their voicemail. I got egg on my
face for this one. I changed that setting to SIP/IAX and packets now
arrive and go where they should. Two problems remain.
1. Still can't
2023 Jun 21
2
Asterisk not replacing private FROM ip with public IP in INVITE
I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction:
From: "MYNAME" <sip:16667778888 at 172.31.253.4>;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4
The IP address above is an internal/non-routable IP, so Twilio is rejecting it. For some
2015 Mar 07
2
AWS/EC2 server selection
Hi Jeff
Are you aware of any challenges of hosting it on AWS? It will help me to
work out alternate plan. Is there any recommendation? Should I split it
to multiple instances and balance traffic across multiple small server
instances? I can use Kamailio to balance traffic.
I see many posts referring to AWS deployment. Please help me to choose
AWS server instance.
*Thanks & Regards,*
2023 Jun 21
1
Asterisk not replacing private FROM ip with public IP in INVITE
type=endpoint
rewrite_contact=yes
force_rport=yes
rtp_symmetric=yes
On 6/21/23 14:36, TTT wrote:
> I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction:
>
> From: "MYNAME" <sip:16667778888 at
2023 Jul 02
1
Get channel variables via ARI/AMI
>> You use the AMI action Getvar[1] which allows channel variables and dialplan functions.
>> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar
I actually tried that, and although I get “success” I never get useful data. For example:
action: Getvar
actionid: act1
channel: PJSIP/Twilio-NA-W-2-In-00000025
Variable: channel(pjsip,call-id)
2023 Jul 03
1
Get channel variables via ARI/AMI
The uppercase command made a difference. I now get a call-id as show below. However, does the call-id look valid? The @0.0.0.0 seems strange.
action: Getvar
actionid: act1
channel: PJSIP/Twilio-NA-W-3-In-00000028
Variable: CHANNEL(pjsip,call-id)
Response: Success
ActionID: act1
Variable: CHANNEL(pjsip,call-id)
Value: 4decf884e3ae74595906283a74f7154e at 0.0.0.0
As well,