similar to: How to Integrate Twilio With Your Rails 4 App

Displaying 20 results from an estimated 200 matches similar to: "How to Integrate Twilio With Your Rails 4 App"

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
2015 Dec 02
4
Issues with Twilio number incoming call and context matching
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 from a non-PBX connected E164 number (a landline), say 555-666-1212. My Twilio number is
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091230/4829ae70/attachment.htm
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
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
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
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 Mar 30
0
WaitForSilence NEVER detects silence,,Post
I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is
2015 Mar 30
0
WaitForSilence NEVER detects silence
I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is
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
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
2023 Jul 01
1
SetCallerPres command gone
I should have included the debug output: <PJSIP/Twilio-NA-W-3-In-00000006>AGI Rx << CALLERPRES(allowed) <PJSIP/Twilio-NA-W-3-In-00000006>AGI Tx >> 510 Invalid or unknown command -----Original Message----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of TTT Sent: Saturday, July 1, 2023 11:37 AM To: 'Asterisk Users Mailing List -
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]:
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
2002 May 17
2
read.table
Hi, I have a data file with columns separated by ";" I read this file without any problem using read.csv2( ) but I had problems trying to read it with read.table( ... sep=";"). So it is not a problem for me, but I wonder if there is a bug here. drt <- read.csv2("t.txt", header=TRUE) # ok dcs <- read.table("t.txt", header=TRUE,
2023 Jul 01
1
AGI script commands
I have an AGI script written in PHP that worked great with Asterisk 13. I'm porting it to an Asterisk 20 site and have a strange problem. I tried running the script from the command line and it works fine; I see the script commands written to stdout like VERBOSE "SmartScreen v1" But when run from asterisk the CLI shows: [2023-06-30 15:50:47]
2007 Feb 28
0
Deriving controller/action from a RESTful URL
If I have a URL/PATH generated via Rails routing: edit_foo_path(foo)) => "/foo/1;edit" ...is there some way that I can take that string and get its component parts (controller, action, id, etc)?? I''m specifically looking for a solution that doesn''t involve me coming up with a regex. I want to rely on code with Rails (which clearly must exist). Thanks! Carter
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