similar to: asterisk pjsip as voip client with multiple registrations

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:

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

Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk]

07.03.2015 0:24, Kevin Harwell ?????: > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in > pjsip. > > I have a

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

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

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

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 @

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

Thanks George I originally didn?t have the 1002@ for the identify. Changed that when things were not working. I changed it back. Unfortunately, the system I am connecting with doesn?t seem to support the line support. Looking at the SIP packets, I see Asterisk send it. Unfortunately, they do not send the line information as part of the INVITE. I checked with some developers of that system

I have a trunk set up for the DID from my provider: [my_provider] type=registration outbound_auth=my_provider server_uri=sip:sip.example.com client_uri=sip:my_username at sip.example.com retry_interval=60 [my_provider] type=auth auth_type=userpass password=123456 username=my_username [my_provider] type=aor contact=sip:sip.example.com:5060 [my_provider] type=endpoint context=from-my_provider

I'm being told by my ITSP that my Asterisk shouldn't be challenging their system to authenticate (i.e. a 401 response) when they send me a SIP MESSAGE (or I suppose a SIP INVITE for that matter). But I'm not sure what a pjsip.conf configuration for that looks like. How does one associate an incoming call/message with an existing authenticated outgoing registration so that Asterisk

Thank you George. I will pass along the rfc information to those responsible for the other switch. I missed the match_header addition to Asterisk. Unfortunately, the only header field that seems appropriate is the To header. On a separate box I am now trying to configure the endpoint recognition. Planning on multiple endpoints to the same switch, so I am trying to use the match_header field.

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:

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.

Hello. I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip. Now i am transfering all from chan_sip to chan_pjsip. And have a lot of questions. First of... system: Asterisk 13.2 on slackware 14.1 Errors on outgoing call: [2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778 request: Failed to create outgoing session to endpoint 'srv_d228' [2015-03-03 00:18:58]

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 Saturday, April 18, 2020 5:42:11 PM CEST Joshua C. Colp wrote: > On Sat, Apr 18, 2020 at 8:47 AM hw <hw at gc-24.de> wrote: > > Hi, > > > > how do I make a bug report? I filled in the form to make a report and > > https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues > > reported by me. > > If successful then JIRA will redirect

Hi all, I have an issue with asterisk 13 and pjsip. I guess it is somehow Firewall related, but I'm unsure. A registration to Sipgate is established successfully: <Registration/ServerURI..............................> <Auth..........> <Status.......> ==========================================================================================

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple