similar to: Looking for the carrier that owns a particular DID

All; I am trying to build and install certified Asterisk 13.13 cert3 on a Ubuntu 16.04.2 LTS host without much success. I am getting the following errors when I try to compile. [CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o res_pjsip/config_transport.c: In function 'transport_apply': res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg

Hey; Thank you very much. I was able to install asterisk from your link. One other question. How are you starting asterisk? Do you use an init script or systemd? Do you think that you could share the script you use? Thanks Again; John V. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent:

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

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

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

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

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 >

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]:

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 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

All; I'm currently running Asterisk 1.8.15-cert7 and am using realtime to store my voicemail configuration. The voicemail application works fine, but the problem I have is that the 'Directory' app cannot find any entries because there are no entries in the voicemail.conf file. When I add a context and an extension entry in voicemail.conf, it works the way it should. Is there

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

    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

Is there a way to tell if a particular fax was transmitted by T.38 or G.711? I'd like to save the information in the CDR, but the "fax show stats" command only gives a summary since the last restart. It would be nice if the FAXOPT variables contained this information but it doesn't. Thanks; John Tech Support Tech Support VoIP Business Solutions 240-215-3479 (Work/Fax)

All; I'm having a problem sending an outbound fax using Asterisk-1.8.15-cert3 and the spandsp fax module using a SIP trunk. I'm seeing hundreds of these: ERROR[14423]: udptl.c:294 encode_open_type: UDPTL (SIP/runcentral_outbound-00000074): Buffer overflow detected (59 + 134 > 175) Has anyone ever seen this before? I have the following configuration. udptl.conf:

All; I'm working on a project (using Asterisk 1.8, but 11 would probably work just as well) where so far I've been able to originate over 1,000 concurrent outbound faxes. I have no problem with that so far. Where I have the problem is that Asterisk is dumping core after the faxes are sent. Now two things happen after the faxes are sent. (1) A fax log similar to a CDR is written to

All; I have a few customers that do a lot of faxing, both inbound and outbound. Some use the Spandsp and some use the Digium FFA modules. What I would like to do is when an outbound fax fails, determine whether the remote end was a fax machine or a plain old phone. I would also like to determine that in the dial plan if I could. Any insight at all with this would be extremely helpful.

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

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA&SIP/deviceB&SIP/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension to dial in order to modify the dialplan. Here is what I