similar to: pjsip: don't require authentication from remote i register to

On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote: > you can try line functionality on the outbound registration which > may or may not work[2] (requires the upstream to adhere to the RFC, > which not all do). My provider seems to implement this. However even with the line=... in the: SIP to address: sip:5555551212@<my_IP_address>:5060;line=dpnlyiu res_pjsip is still

Hello, We have a provider which is using Kamailio as front end. Our asterisk 13/chan_sip server has no problem to register and pass/receive calls form this provider. Now we want to move to asterisk 16/pjsip and face problem. Registration is OK but when we pass a call our INVITE never receive answer from the provider. We opened a ticket to their support but in the mean time we want to know

hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:308 at example.com:5060 client_uri=sip:308 at example.com:5060 [308](auth-userpass) username=308 password=pass [308](aor-single-reg) contact=sip:example.com:5060 [308](endpoint-basic)

I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic configuration works, and I am connected to a SIP trunk using SIP.US, and have set up my inbound calling which works correctly (when I call my PBX DID, the call does come into my PBX network). The issue is that I am not able to make outbound calls, because the call fails with the error:

Hi all, In my case I using realtime, here is how it looks in plant [10001] type=registration transport=upd_static outbound_auth=10001 server_uri=sip:600 at 192.168.1.1:5060 client_uri=sip:600 at 192.168.1.4:5060 [10001] type=auth auth_type=userpass password=600 username=600 [10001] type=aor contact=sip:192.168.1.4:5060 [10001] type=endpoint transport=upd_static context=dialmap disallow=all

That was the issue, thanks. I now am able to get the caller ringing on an outbound call, but an external phone number (E164) I am dialing does not ring. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com > wrote: > > > On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I have setup my

I've seen this happen three times in the wild now.  I've been trying to isolate the source of the issue, but so far it seems like there's not enough debug output to know why this occurs. Long story short: - Start Asterisk - PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk is behind NAT).  SIP is handled correctly, Asterisk responds OK with RTP media address of

Hello. Asterisk 13.2, PJSIP. Problem: I do not get any AMI events when changing the status of the contact. When using chan_sip I got "peerstatus" event. When using res_pjsip and devices (endpoint configuration) I got "peerstatus" event. When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION i got "registry" event. When using

On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote: > On 4/5/19 10:36 AM, sean darcy wrote: > > I'm trying to set up pjsip to work with an obi202 and google voice. But > > I can't configure the endpoint. > > > > pjsip: > > > > [obi202-auth](!) > > type = auth > > auth_type = userpass > > password = <mypass> > > >

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1&t=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate it all here, I've posted my logs and conf files on that thread, too. Problem is that while

Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character '@' [2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 aco_process_var:

Currently using PJSIP. First, they want me to get this working with the existing PJSIP configuration, but then setup a second box using chan_sip performing similar work. For PJSIP... I currently have an endpoint configured to a system using IP based authentication. It is configured with a match setting in the endpoint section. All channels coming from that IP address go to this endpoint. They

Yes, I think the dial does get executed (sonny calling outbound 202-555-1212): core set verbose 3 Console verbose was OFF and is now 3. -- Executing [912025551212 at from-internal:1] Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from "" <sonny> to 12025551212 through fromgw") in new stack [Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @

On Fri, 2019-03-01 at 15:41 -0500, Joshua C. Colp wrote: > > I don't understand what you mean. Your ITSP has stated that they > don't want you to do authentication with them, so you can't. They are implying, as I am understanding them, that somehow SIP packets they send me shouldn't need to be authenticated because they are associated (i.e. "identify"ed in

I am trying to configure a connection to BluIP. I am able to make incoming calls work. However outgoing calls are not working. For the Outbound Registration, I noticed the contact field is always the internal IP address of my pc instead of mycompany dot com I can Originate (using AMI) to my Vitelity trunk (IP based authentication). However, when I Originate to my BluIP, it is being rejected.

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there.  I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds.  This happened while in the automated attendant area.  This problem did not occur with Asterisk 13 LTS.  I reverted 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]:

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

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

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