similar to: pjsip trunking configuration issue

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

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

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

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 >

    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

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

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

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]:

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

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

Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump on asterisk server showing UDP packet bound for my remote endpoints internal IP: 17:07:57.130212 IP

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

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

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

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

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

>> 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)

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,

On Sun, Jul 2, 2023 at 4:39 PM TTT <lists at telium.io> wrote: > >> 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

Check out Twilio From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Federico Sent: Thursday, August 17, 2023 11:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users at lists.digium.com> Subject: [asterisk-users] Question about Sip Trunks who support Stir Shaken I am looking for a decent provider of SIP