Freepbx failed to authenticate on invite to. 0 VICIDIAL TRUNK [FREEPBX] disallow=all .
Freepbx failed to authenticate on invite to I’ll be you’ve got everything set up for one and are using the other on port 5060. I believe the problem lies there. 220) SIP Trunked to PBX2 FreePBX 2. Thus not a sip->pjsip conversion issue. 201. 71) Ext 6203 and 6204 On PBX1 (CUCM) the ext 5222 has been set to forward all calls to ext 6204 Only one calling scenarios does not work - if a user on PBX2, such as 6203, calls ext 5222, the forwarded call goes to PBX1, an invite comes back to PBX2, but is being rejected as unauthorized. 211. Here is configuration: host=192. CA. c: 673log_failed_request: Request ‘INVITE’ from ‘“+0xxxxxxxxxxx” sip: [email protected Jan 26, 2022 · Hi, on my FreePBX 16. c: Request 'INVITE' from '<sip: [email protected]>' failed Dec 11, 2013 · Hi, I’m trying to use one FreePBX with Asterisk 1. There is a bogus newline, between “cn” and “once”, in your log of the header, as sent by the phone. Only the single old one, which initially created, work perfectly from the At first glance, it would appear that the incoming call was challenged for authentication, and that 200 then failed to authenticate on the second INVITE sent. c: Request ‘INVITE’ from ‘<sip: [email protected]>’ failed for ‘147. 1~dfsg-1+deb10u1 and FreePBX 15 (framework 15. 46:5060’ (callid: 848a8d68 Jan 24, 2020 · Was the carrier trying to send a re-INVITE ? You need something like “sip set debug on” then post pertinent logs… May 14, 2018 · Oh I believe you. c:676 log_failed_request: Request ‘OPTIONS’ from ‘ sip: [email protected] ’ failed for ‘yyyy:5060’ (callid: 42f56ad8-e568-4292-bbbc-ad2b411996e6) - Failed to authenticate The trunks are set up as follows: PBX1: Dec 20, 2011 · I have setup a connection between FreePBX and Vicidial, I can call from FreePBX to Vicidial with not but I can’t get inbound calls from Vicidial to FreePBX. 135. Any help on getting the inbound issue resolved out be appreciated. 6. Oct 26, 2016 · I really do have identical problems. 243:56120’ (callid: 79lZFiJyOl) - Failed to authenticate This happens for all new created pjsip extentions. 16. 22. c: Request 'INVITE' from '<sip: [email protected]>' failed for '192. 32:2053’ (callid: 630c8c01f282-zgnou2gwgobx) - Failed to authenticate On the Phone set it is showing that not registered: Test I am newbee could you please guide how to fix this ? Dec 11, 2024 · I can capture it using wireshark/tcpdump, but the packet arrives at FreePBX and is rejected due to “authentication failure”. Jun 29, 2020 · It sounds like your provider is using IP Authentication to send the calls to your machine. They mostly work with Avaya and are not sure how to help. 8 (IP address 192. 8) last night and setup my PBX. 1 failed to authenticate Jun 30, 2021 · Hi Ive migrated from SIP to PJSIP using the tool. I am havin gissues with my freepbx server. However when I try to register a line and connect to FreePBX it does not register. cynjut (Dave Burgess) August 30, 2016, 1:34pm 2 Hello, I've recently reinstalled FreePBX into a virtual machine after decommissioning my physical PC. I can ping the PBX server or phone from each end. It is only for family and home business use so not business critical as I have all calls forwarded to my mobile through my trunk provider. c:25904 handle_request_invite: Failed to authenticate device sip: [email protected]:5060;tag=d11f24229f52d2472c23756e2e0effbc i don’t understant why because right now i don’t have any sip or device with number 1001. 75 on a Raspberry Pi 3B. Sep 8, 2015 · To gain full voting privileges, this problem came up when i tried forwarding calls. c I have some Yealink T-46U phones that connect to internet and get IP address. If that’s the case, you need to look at what IP addresses the calls are coming from. The extension and the password are the same as I setup in the PBX. All devices are using pjsip (and had done so before migration). 1 (10. Oct 27, 2024 · SIP communication uses standard SIP registration with authentication credentials. I am attempting to use this as a local service acting as phones for an office like setting. 7. If I May 11, 2020 · [2020-05-10 20:00:37] NOTICE [9434] res_pjsip/pjsip_distributor. 0 VICIDIAL TRUNK [FREEPBX] disallow=all Aug 30, 2016 · Yet when I try to make a call I get: handle_request_invite: Failed to authenticate device "22" I’ve been hammering at this for two days and I’m going insane. 0. 71. NOTICE[6242]: res_pjsip/pjsip_distributor. I am running FreePBX 15. I would try to create a new pjsip endpoint, which may be either listening in port 5160 or on port 5060 is sip settings are “both”. c:676 log_failed_request: Request ‘REGISTER’ from ‘ sip: [email protected] ’ failed for ‘91. 177 is the IP address of my HA node, not the floating IP, which should be the address in the invite packet. (4 Mitel/Astra Dect, 1 Grandstream WLAN, 1 Grandstream Video, 1 old Snom 260, 1 Softphone) Registration and connections to and from my providers (so any external connections) are working properly. 223 secret=1111 type=peer insecure=port,invite . . On my other machine, same version of Asterisk, I have added SIP Trunk registered to that extension (200). Registration is OK Oct 13, 2023 · I am trying to find why I can make outbound calls, but inbound fail to authenticate. Problem Description: When the router boots up, it appears to initially form a SIP trunk connection to the FreePBX server but fails shortly afterward. Internal through my PJSIP extensions. c:676 log_failed_request: Request ‘SUBSCRIBE’ from ‘sip:MAC%3A000B82A62107@224. I installed it twice without changing anything, I just created the extension and tried to authenticate with MicroSIP. I bought two Cisco SPA504G phones cheap on eBay just to try things out. 49) and only the first endpoint succeeds in registering. These are the configuration setups I came up with: Trunk #1 disallow=all username= type=friend secret= qualify=yes Apr 15, 2020 · [2020-04-15 10:39:40] NOTICE [2745] [C-000000b1]: chan_sip. 10. But i cannot get some extension to register res_pjsip/pjsip_distributor. Looking for some ideas Thanks in advance. Once I Jul 12, 2022 · PBX2: NOTICE [28732]: res_pjsip/pjsip_distributor. 4. Any help is greatly appreciated! For a school. This can be an external firewall redirecting specific hosts traffic to your SIP port or the Integrated Firewall blocking traffic from any place that you don’t want traffic to come in from. Mar 5, 2017 · I am getting a “Failed to authenticate on INVITE to sip: [email protected] ”, which is strange, because 10. xx fromuser=99xxxxxx Jun 17, 2014 · I just downloaded the distro 5. Now all my phones fail to authenticate. 223 username=200 authname=200 fromuser=200 fromdomain=192. 8. Aug 2, 2022 · Is your SIP-channel-driver in settings/advanced setting on “both” or non pjsip only? Can you find the new extension at admin / config edit → xxxsip. 75’ failed for ‘10. 1. I have configured two non-identical trunks with different parameters for testing purposes on asterisk (11. x. 168. 95:5080’ (callid: 1559319419-5080-1@BA. 90. Then failed to authenticate. At the remote end, is 50. BA. The actual problem here is that the endpoint 200 does not exist within Asterisk. Jul 18, 2025 · NOTICE [10696] res_pjsip/pjsip_distributor. [2022-08-03 18:34:34] NOTICE [62603]: res_pjsip/pjsip_distributor. Any help on this? Jul 14, 2022 · I can access their GUIs both from a Browser and from the phones GUI and try to re-register them but they will not register and I get “Failed to Authenticate” errors on them. c:2430 authenticate: 127. When i attempt to register it gets denied and stated no matching endpoint found. 5. xx. I restarted all the system but this notice persistes. I tried disabling the firewall and adding ignoreip in fail2ban. 16:12615’ (callid: 866720488) - Failed to authenticate SIP Trunk is with RealmConnect in Georgia. 87:63464’ (callid: 320061365-2098964229-320328617) - Failed to authenticate That’s an odd extension. I have one provider but must have multi SIP trunk’s because of billing (different users etc. 1(1. Nov 11, 2022 · I setup a freePBX iso to connect to 2 asterisk servers (stock asterisk 16 and 18). endpoint. Thanks for any help. Jan 15, 2024 · I am completely new to FreePBX and trying to set things up for the first time. ms but it is always the same. Mar 26, 2024 · I have successfully setup my freepbx distro, and extensions are registered successfully on my LAN and working fine However, intermittently some of the phones spit these errors in the CLI and my fail2ban blocks them. The inbound traffic made it from the outside, so the network appears to be OK. 223) to simulate SIP PBX, so I created there extension 200. c:2467 authenticate: 127. Feb 3, 2025 · The “initial” request contains an Authentication header, so can’t actually be the initial request. Jun 20, 2014 · Hello. 6>;tag=as08b7e6b9 for INVITE, code = -1 pcap has: [Expert Info Sep 23, 2010 · 2010/9/27 update I knew this was going to hurt when I found it … The problem turns out to be a difference between what you can use in FreePBX/Asterisk and the older version of Asterisk we had running: FreePBX only allows you to use the extension number as the UserID / Authorization name [in Bria terms]. I have included my SIP Debug from both Vicidial and FreePBX. admin2all (admin2all) August 3, 2022, 12:44pm 43 Mar 23, 2017 · When I create my extension from the FreePBX create new SIP extension and try to connect afterwards I get Forbidden on my SIP client. Jan 17, 2019 · Also, I still receive the same error "Failed to authenticate on INVITE’ when attempting an outbound call regardless of combination. Check your port assignments in the Advanced SIP settings tab. 18) . Apr 3, 2019 · Hi ! I have problem with Multi SIP trunk registrations when I set caller ID on trunk and call DID on second Trunk on same FreePbx. Inbound calls get: NOTICE[3947][C-00000009]: chan_sip. The phone is on a vlan on the network. 1 tried to authenticate with nonexistent user ‘cxpanel’ [2014-06-16 19:13:48] NOTICE[14437]: manager. 65-14 (1. 0) SDP Session Name: Asterisk PBX 1. Everything seems to be working, however when I go to the CLI I keep getting this notice [2014-06-16 19:13:48] NOTICE[14437]: manager. JF) - Failed to authenticate Also when i am trying to make a call i am getting res_pjsip/pjsip_distributor. i can make outbound calls with no issue but the problem with inbound calls I see the log message: NOTICE [2419]: res_pjsip/pjsip_distributor. Jun 10, 2023 · Otherwise, when incoming call is sent via FXO, FreePBX will reject the call with res_pjsip/pjsip_distributor. 60. This is how my (test)… Nov 22, 2016 · Do you have PJ-SIP assigned to port 5060, or Chan-SIP? IP authentication only works with Chan-SIP (last I checked). I have tried changing the password a number of times on VoIP. c:676 log_failed_request: Request ‘SUBSCR IBE’ from ‘sip:201@10. Sep 27, 2024 · FreePBXEndpoints freepbx, configuration dawoud (dawoud) September 27, 2024, 11:15am 1 i setup freepbx trunk that is connect to gsm gateway from dinstar. 20. Aug 29, 2022 · Request ‘REGISTER’ failed for 192. c: Request ‘REGISTER’ from ‘“1001” sip: [email protected] ’ failed for ‘103. c: Request 'INVITE' from '<sip:[Incoming phone number]@[FXO IP]>' failed for '[FreePBX IP]:5062' (callid: [FreePBX Call ID]) - Failed to authenticate. We sell freepbx and pbxact at work and we have in house production and test systems that are not exhibiting this problem that have pjsip extensions I just would like to know if there is something i could try on my system before i decide to give up and nuke / reinstall. 43 I notice this messages: 2022-01-26 17:49:56] NOTICE[1303] res_pjsip/pjsip_distributor. Aug 19, 2024 · I just upgraded to 17. I'm not really sure where to go from here, as I can't really find a lot on Google about these INVITE failures -- everyone seems to be trying to talk to a commercial SIP provider, so I'm not sure if the error is in my FreePBX instance or in the settings on my phone. 46’ failed for ‘10. 0) with Elastix (2. conf. 100. I am only able to receive calls on my main number, any other numbers fail to register at the INVITE stage and I am not receiving any DID from the trunk so Hi everyone, I installed FreePBX from source yesterday, and I can't figure out why Linphone won't connect to the server. 0 x64) on a Linux based machine (x86_64; 2. Smooth upgrade, no issues. ) and have setup: Peer details: username=99xxxxx type=peer sendrpid=yes trustrpid=no secret=xxxxxx qualify=yes insecure=port,invite host=xx. 1:55741' (callid: 1921743871-1471166675-1252888528) - Failed to authenticate 171957[2022-01-26 17:49:57] NOTICE[1303] res_pjsip/pjsip_distributor. No outbound calls. This is one thing that bothers me about freepbx / pbxact, you get this one off issue and if you don’t pay Aug 25, 2023 · Hello, I am very new to the freepbx platform. User Agent: FPBX-2. 2. None of them worked out (when I try calling these numbers it doesn’t even connect to them). x a firewall (is the phone Jul 9, 2020 · In general, you do not want your SIP port exposed to the Internet without some kind of prophylactic measure in place. 11 (10. I cannot get them to register and having looked at other posts on your forum addressing similar issues, can’t get an answer. The older Asterisk we were using allowed text for UserID / Authorization name. Two extensions (21 & 22) are registered on the server using Mar 3, 2015 · Ext 5222 on PBX1 Cisco CUCM9. How about from the provider’s side, what could be causing this? But a few restarts later, I realized that the issue is even stranger: I setup an Asterisk server with Debian 10, packaged Asterisk 16. c:19648 send_check_user_failure_response: Failed to authenticate device <sip:417xxxxxxx@208. 176. zrm8iwjumjmyguikqvopdxd96wnr8lrhxwto1eb7c6s5aopz