search for: twilio

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: Cal...

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 en...

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 T...

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 add ;transport=tcp to my Origination URI after wireshark revealed everything was received as UDP into Asterisk, so we can rule out that...

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 > logging I notice that the outgoing invite...

...ind 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 that mean the initial invite will contain authentication info? Or does Asterisk still wait for a 407?? (I'm wondering if maybe Asterisk is working normally, this is a Twilio config problem). And I confirmed the CID info matches an account on...

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...

*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...

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]: r...

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 leave the VoIP > island). > > Here's what I have right now in pjsip_wizard.c...

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

    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 > >...

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/...

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

...aitForSilence(5000,2,60); > AGI(agi://127.0.0.1/playmessage,${CALLID}); > AGI(agi://127.0.0.1/saytext,"Goodbye."); > Hangup(); > } And the CLI just outputs: > == Using SIP RTP CoS mark 5 > > Channel SIP/twilio-0000006e was answered > -- Executing [100 at makeCall:1] Answer("SIP/twilio-0000006e", "") in > new stack > -- Executing [100 at makeCall:2] > WaitForSilence("SIP/twilio-0000006e", "5000,2,60") in new stack > -- Waiting 2 time(s)...

...aitForSilence(5000,2,60); > AGI(agi://127.0.0.1/playmessage,${CALLID}); > AGI(agi://127.0.0.1/saytext,"Goodbye."); > Hangup(); > } And the CLI just outputs: > == Using SIP RTP CoS mark 5 > > Channel SIP/twilio-0000006e was answered > -- Executing [100 at makeCall:1] Answer("SIP/twilio-0000006e", "") in > new stack > -- Executing [100 at makeCall:2] > WaitForSilence("SIP/twilio-0000006e", "5000,2,60") in new stack > -- Waiting 2 time(s)...

...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] VERBOSE[1264031][C-00000025] pbx.c: Executing > [s at function-smartscreen:2] EAGI("PJSIP/Twilio-NA-W-3-In-00000068", > "smartscreen/smartscreen.php,"GEORGE SMITH" <+1234567890>") in new > stack > > [2023-06-30 15:50:47] VERBOSE[1264031][C-00000025] res_agi.c: Launched > AGI Script /var/lib/asterisk/agi-bin/smartscreen/smartscreen.php > >...

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 A...

...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 reason Asterisk is not replacing the private IP with my public IP address. My pjsip.transport.conf contains a stanza for this transport with: local_net=172.31.0.0/16 Is that all that's needed for Asterisk to replace the from IP with the external IP? I'm not cl...