Asterisk register failed. xx:5060' - Wrong password .

Asterisk register failed See the Inbound Registrations section for details on What are the possible reasons for a SIP register failure? I have setup an Elastix box (asterisk/freepbx based) and added a trunk for an external VoIP provider. I have my FreePBX 16. I have configured the sip. Hi, I setup a freePBX iso to connect to 2 asterisk servers (stock asterisk 16 and 18). When the phone gets blacklisted what is the reason, will show you in the blacklist description. To make finding the problem even harder, outgoing calls worked as expected because each call does its own registration separated from the inbound registration. However, the phone couldn't register with Asterisk. xx. xx:5060' - Wrong password I have setup an Elastix box (asterisk/freepbx based) and added a trunk for an external VoIP provider. While I have traditionally done asterisk cli wise, registrations usually are not a problem. I’ve been at this for three days now and cannot get these phones to register. Products LINKVIL Wireless UC Products SIP Phones SIP Security Industry Products 2-Wire Products Business Conferencing Headset Service Platform Accessories EOL Products Solutions 3CX Solution Does anyone have tips for where I can look for logfiles to debug failed registration either on the handest or on the PBX/SIP server? Current Asterisk Version: 13. Configuration File: pjsip. You'll likely see causes of a 403 forbidden on a register attempt? Other providers work fine using the same syntax. When I try to register it shows fetching registrations failed: 503 Service Unavailable. Johansson - at the CLI enter SIP DEBUG (and SIP NO If that’s not your issue, please paste the Asterisk log for the failing registration (with pjsip logger on) at pastebin. Failure Events: Off T. Basically, FreePBX 2. 83. You propably want to use chan_sip OR chan_pjsip. txt. The phone is on a vlan on the network. You can check that device added in sip_additional. c:28627 handle_request_register: Registration from '<sip:[email protected]>' failed for '10. peer settings: [remotepeer] type = peer host = dynamic insecure = port,invite context = remotepeer-Inbound directmedia = no dtmfmode = rfc2833 callcounter = yes nat = no contactpermit=1. It is already set to no but though fails, still processes the INVITES from unauthorized users. Device sends the register message with nonce value with call id 4EdqB3YwpQHIYGPtj9c and CSeq no. 1). It can be caused by: Incorrect credentials: Ensure that your SIP trunk username, password, and domain Rejected: Asterisk attempted to register but a failure occurred. This is the ttyIAX1 file contents on the fax server: [root@dmxfax]# cat /etc/iaxmodem/ttyIAX1 device /dev/ttyIAX1 owner uucp:uucp mode 660 I have been battling with some phone issues for quite a while and I’m looking for some ideas to troubleshoot. Username - The username portion of the registration. ChannelType - The type of channel that was registered (or not). com SIP/2. and anyway I'm trying to diagnos a SIP client failure. The same failure mode occurs with realm defined, but not set specifically to the realm value from the challenge. When an INVITE comes, asterisk tries to check the given user Outbound Registration. Confirm that you have turned off FreePBX Firewall for testing, or post the output of iptables -vnL Hello, Yealink T42G, latest firmware. conf'. and they say nothing is wrong. But not able to register it. conf file and in asterisk via. I downloaded the zoiper on the same system where i setup the asterisk now I want to redirect flooder IP address with messages like "Failed to authenticate device" to my perl script. Anybody got a clue? david55 Note that your log is showing the registrations succeeding on Stack Overflow for Teams Where developers & technologists share private knowledge with coworkers; Advertising & Talent Reach devs & technologists worldwide about your product, service or employer brand; OverflowAI GenAI features for Teams; OverflowAPI Train & fine-tune LLMs; Labs The future of collective knowledge sharing; About the company Visit the blog 文章浏览阅读3. conf that comes with a make samples-- defaultuser is described as "Authentication user for outbound proxies". sudo netstat -tulpn. The issue that I’m facing is I can’t get any softphone (zoiper in my case) on an external network to connect to the box it just says registration failed forbidden 403. After that, the sip show peers command should return some kind of status. In a setup with an asterisk SIP server (tested with version 13. Stack Exchange Network. Scenario is regarding SIP register on LTE network. 20. 5. freepbx-centos*CLI> pjsip show registrations Information sur ma configuration: Debian 7 J'ai un debian compilé avec FreePBX, mais je m'en sert pas, je configure directement via asterisk dans sip, extensions, users, voicemail, etc. I have reported it to them each time and by the time they reply it had started working. username - Matches the endpoint or AOR ID based on the username and domain in the From header (or To header for AORs). Asterisk SIP. 7001 and 7002. 1. asterisk作为sip注册服务器，保存客户端注册信息。下面分析从sip协议栈启动到第二个功能的具体实现：sip协议栈作为可加载模块在系统启动时 Hello I am new to asterisk doing first time configuration and setup. I have gotten Yealink T46U phones. 10. txt core-asterisk-running-2024-10-21T07-18-40+0200-brief. 106:49234’ - W However when I try to register a line and connect to FreePBX it does not register. Viewed 2k times 1 . Unless otherwise mentioned, everything is at defaults. Check modules. 200. 850 & Q. Happy to pull and post any info necessary, but really looking for tips on how to figure out where the failure is and why. c:28059 handle_request_register: Registration from '"pushpa" <sip:[email protected]>' failed for '116. 203. Back. 10: 34: March 4, 2025 Inbound calls fail on PRI line. 1/32 permit=1. jpeg ( 2) connection_shema_001. conf [general] transport=udp [friends_internal](!) type=friend host=dynamic context=from-internal disallow=all allow=ulaw [demo-alice](friends_internal) secret=verysecretpassword qualify=yes ; put a strong, unique password here instead qualify=yes [demo-bob](friends_internal) secret=othersecretpassword ; This tool watch for REGISTER request with "Failed to authenticate" in 'journald' for unit 'asterisk' and ban unwanted IP with iptables + ipset (and provides metrics for Prometheus monitoring). pdf. I am not using it for internet or behind NAT router. Commented Sep 11, 2012 at 13:01. 3) to the asterisk server 2 which is in the other network (ip:192. Then, if you have firewalls, then you now have to open that non-standard port too. When I first register sip client with Asterisk it works fine but after one or two hours some (not all) starts to fail registration and counter shows it will re-register after 1500 seconds or some times 765 sec. 168. and registration get successful. When I get to the One of the most common issues in Asterisk SIP trunk troubleshooting is SIP registration failure. freepbx. Registrations: For Asterisk I am scratching my head over this one. But fails when tried from external network. See the above section for more information on failures that may occur. Anyway, probably the best approach to troubleshooting registration errors is to watch the Asterisk CLI as it attempts registration and to see what it says. I'm using Asterisk to register sip client but it shows me this message: NOTICE[3212]: chan_sip. conf file. While it's not a proxy in this case, I believe this is the parameter that will be used First, the register line should have a path set at the end, like: register => myusername:mypassword:[email protected]/84106639 Then do a sip reload or service asterisk restart. 0 403 Forbidden Via: SIP/2. Analysis - The Asterisk CLI You can request technical assistance by searching the knowledge base for information SIP registration failure troubleshooting. I checked to ensure that all After the change, does anything appear in the Asterisk when the device attempts to register? If so, at the Asterisk command prompt type pjsip set logger on and paste the log for a failed registration attempt. conf and added two entries. I have been able to provison 3 handsets manually but further devices are showing as Note that this only tests that Asterisk can receive a REGISTER request and reply back with a "200 OK" response, not whether or not Asterisk has created a SIP peer internally or performed any decisions based on the receiving of the REGISTER request. 16. 0/UDP 69. 25. 11) SDP Session Name: Asterisk PBX 1. On the other side an asterisk with ip 10. 04 just installing Asterisk (apt-get update && apt-get upgrade && apt-get install asterisk). 931 unless specified) MFC/R2 SIP/PJSIP Motif; AST_CAUSE_NORMAL_TEMPORARY_FAILURE: 41. So, since I can't register with the server I Michael, I’m fully aware of this. Checking with pfctl -s state -vv | grep FreePBX-IP | grep :5060 confirms that the states still refer to the old, now invalid public IP address, so that the SIP-UDP packets have a wrong origin IP and the anwers never come back. 38 support: No T. I did as you say, I put Server details in Global Settings page in “Basic SIP Network Settings” section, leaving the “Basic SIP Authentication Settings” empty and filling those details in Line1 page while leaving out the server info there, at least now I have some Account name: Account1 Server: <ip-address-of-my-asterisk-server> User name: guest Password: test Caller ID: guest On the Asterisk server, /etc/asterisk/iax. Modified 10 years, 2 months ago. I know that the phone can register with FWD and I have used tcpdump to see the registration packets arrive at the Asterisk server, but nothing goes back. in, in internal network it is getting registered. And the provider. registry Provisioning method is Local LAN (in the office). When I give the domain name as erss. Rejected. The “header” endpoint identifier: is registered by the res_pjsip_endpoint_identifier_ip. But a few restarts later, I realized that the issue is even stranger: I setup an Asterisk server with Debian 10, packaged Asterisk 16. 1; Network responded back with 401 unauthorized. My end points were set connect to the server on port 6060 which is the port i designated for pjsip in “Asterisk SIP Settings” Comparing to one of my Asterisk-to-Asterisk SIP trunks It looks like what I use is the defaultuser= parameter in my sip. What I have below is the cli output of the various settings. conf to prevent one of them from loading In your CLI it seems, ekiga is registered on asterisk zoiper registration failed. 04. conf register => XXXXXXXXXX@ [email protected] :5060/XXXXXXXXXX How do I setup Proxy and STUN Server configuration in sip. context: (not set) Regexten on Qualify: No Caller ID: Unknown From: Domain: Record SIP history: Off Call Events: Off Auth. txt, realm is undefined. I have a Spectrum Business Class Internet, Ubiquiti router, Ubiquiti Switch for the core, and a number of Yealink Phones. sip registration failure troubleshooting. The qualify=yes option is useful too to check IP connectivity and SIP service status. Registered. 0. The phones are Yealink T27-G. 49) and only the first endpoint succeeds in registering. e. I have loaded firmware 9. Ask Question Asked 8 years, 10 months ago. In case you didn't know this doesn't restart Asterisk or reload anything. Suddently - without any configuration- the indoor only shows the live Picture from the outdoorstation but i cant use it as intercom New IP handsets failing to register Hi guys, I'm encountering an issue when trying to set up new extensions (with new linked users). Аs a first step change your register string like: register => username:[email protected]\Myprovider and then add the outgoing and incoming dialplan in your extensions. 0/24 username = remotepeer secret = remotepeerpass My goal is to make a call from softphone (on windows lite with ip: 192. Between the lines indicating 401 and the lines indicating 200 lie several seconds (~30s). The 'v' is verbose, or display more text. Confirm that no entries at all appear in /var/log/asterisk/full when the extension attempts to register. If you would like to refer to this comment somewhere else in this project, copy and paste the following link: I am getting a lot of failed authentications in my asterisk logs. c" after: ast_log(LOG_NOTICE, "Failed to authenticate device %s [IP: Outbound Registration. I have a cloud hosted pbx with FreePBX managed by a 3rd party. 19 (Asterisk version 13. When I try to log in from my Joshua Colp is the Asterisk Project Lead. Cisco 7960 and 7940 phones (a fair So my guess has to do with this part you wrote: "asterisk -vvr". 9 on Centos (5. I logged into the phone gui by the ip address. To work seamlessly I'm new at asterisk and following asterisk example: sip. Internal through my PJSIP Applicability Firmware version: Any Model: S-Series Problem Description SIP trunk registration failure. 100. Stopped: The outbound registration has been removed First, the register line should have a path set at the end, like: register => myusername:mypassword:[email protected]/84106639 Then do a sip reload or service asterisk If you're having troubles getting a phone to register to Asterisk, make sure you watch the Asterisk CLI with the verbosity level set to at least three while you reboot the phone. If not, what if anything appears in sngrep? If nothing there, either, paste a new log from the phone. jpeg Description: Hello, I have problem with registration SIP trunk using chan_pjsip. confthere should be a section for your I am using a local asterisk server on a machine behind a router. 7 rear a FW (NAT inside) with the public static ip sip. I am using SPA 8000 as 8 SIP Clients directly connected to Asterisk Server over LAN. I have Okay, been here for a while. After the first irector left, they didn't use the telephone and Issue 1: SIP Registration Fails. Asterisk restarted resolved the issue. Temporary failure a SIP peer has not registered or sent a REGISTER request with an expiration of 0. If Asterisk show that your softphone is unreachable then you have to check the path from your softphone to the Asterisk to find where the SIP packets are getting lost. Paramahansa Werner Polo Vieyra Thu, 21 Oct 2004 09:13:49 -0700. that I am having sip registration problems with works fine with iax. The asterisk-server has a fixed local IP and the routers external IP is registered with dyndns and can be resolved from outside my network. Hi, I am relatively new to Asterisk/FreePBX, so I am probably missing something extremely basic here. 5 and asterisk 11 with freepbx 2. 文章浏览阅读533次。本文来自 csdn lidp ,转载著名出处，谢谢。对于注册功能，asterisk sip协议栈提供两种服务，1. 26. 2022-05-19. ve@proves. 74. This is the config for one of the extensions: [11] In settings > Asterisk SIP Settings > chan_pjsip > (udp) Port to Listen On = 5060 (default) In SIP Network I set it to my network, which is a /16 of public IPs. I can ping the PBX server or phone from each end. I add in "chan_sip. Clicked confirmed at bottom and after a few moments shows Hi all! I have a serious problem with connecting the DS-KV8213-WME1 and a DS-KH6320-WTE2 as indoor station. txt core-asterisk-running-2024-10-21T07-18-40+0200-full. 3. 133. 13. Allow Anonymous Inbound SIP Calls and Allow SIP Guests are set to no. When it boot up, it display "registering" and it goes off for few miniutes. 0 and 13. registration. 29. I tried everything but no success. Failed. I was using PJSIP extensions but converted them all to Chan_SIP on port 5160. Visit Stack Exchange All things related to Asterisk! Asterisk Community Asterisk Res_pjsip request failed for 'REGISTER' no matching endpoint. If not, what if I'm using Ubuntu 16. 380895 JVM: As soon as the WAN-IP changes, Asterisk is unable to re-register. 85’ failed for ‘192. 1~dfsg-1+deb10u1 and FreePBX 15 (framework 15. Ubuntu + Asterisk = Registration from '"100" failed for '10. Free User Joined Feb 15, 2022 Messages 11 Reaction score 3. registration_custom. 3), we noticed a sort of wait time between the point where the SIP session expires and baresip re-registers. I have had 3 instances in the last 2 weeks when the sip registration fails at the provider and of course the system doesn’t work. thx for answering, wasn’t hoping that somebody might answer and so stopped checking after few days. Device latch with the network and sends SIP register with call id 4EdqB3YwpQHIYGPtj9c and CSeq no. At first, I thought it was an issue with a certain phone. 2 FreePBX 14. 1) I am able to register a SIP client with the server from outside my LAN using the domain name [email protected] Therefore, in order to register your SIP provider with your Asterisk phone system using registration based authentication, you will need your SIP Registrar / proxy, username and password. Intrusion detection is enabled, Firewall is enabled and whether the firewall responsive is enabled or not, I still see a lot of these in my logs. Unregistered. 11 SDP Owner Name: root Reg. 136. Stack Exchange network consists of 183 Q&A communities including Stack Overflow, the largest, most trusted online community for developers to learn, share their knowledge, and build their careers. 17: 26: Register, but no dynamic contact, and inbound call auth failure. g: you use 6XXX series to dial to the provider: I’m new and have freepbx working using e4sip. No outbound calls. Is there a way to completely block these unknown registrations and not show them in the asterisk Registration entry in /etc/asterisk/sip. You could try and reset all the After the change, does anything appear in the Asterisk when the device attempts to register? If so, at the Asterisk command prompt type pjsip set logger on and paste the log for a failed registration attempt. But the problem is in registration between the two asterisk servers which are behind NAT. Network issues: Check your network’s firewall rules and NAT settings. . Thread starter ve@proves; Start date May 25, 2022; Status Not open for further replies. But maybe I am missing something? Anyway, as the subject field says, “peer is not supposed to register” I am trying to register my Sipura ATA 1000 against Freepbx. conf¶ [registration]: The configuration for outbound registration¶ Since¶ 12. Check these points: 1- In sip. conf This is what I put in there and resulted in " Registration Failed" Bildschirmfoto 2022-09-07 um 16. 31:5060;branch=z9hG4bK027fe0e3 CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration Content-Length: 0 SIP/2. Generated Version¶ Unfortunately all my attempt have failed I tried Grep to find the string - no luck; Edited: pjsip. It's an existing system I didn't configure and there are handsets working which were set up prior to my involvement. Visit Stack Exchange [Asterisk-Users] Freshmaker failed register test. I have set up a simple Ubuntu 14. 5 I think - latest release with relevant updates. Here is the logging in the phone : 3378: NOT 10:14:53. 8). I plugged the phones in on my network and I got a IP address for phone that was correct. teliax. conf verify that the line with externip contains the external IP address your Asterisk is using. like:. Does that change anything? I have a Freepbx 15 and Asterisk 17 free distro box that I built for a church and the system for some reason cannot register Polycom VVX310 phones using version 1. (Of course, the two Digium D40’s work flawlessly with minimal to no configuration needed). Registration is COMPLETELY separate from the rest of 'pjsip. Environment: Attachments: ( 0) asterisk_debug_info. It can be caused by: Incorrect credentials: Ensure that your SIP trunk username, password, and domain settings are correct. On January 11 the adsl goes down near 18:20 (i see on a mrtg graph). I am havin gissues with my freepbx server. Back to top . Your DialPlan is not correct clearly from your configuration files. If the device is expected to register, then it may be that the device is either not properly configured or that there was a registration failure. First i got everything setup and it worked just fine. 04 and Asterisk 13. 2- In sip. I have set up a virtual box running centos 6. c:28073 handle_request_register: Registration from ‘sip:1002@192. It is using the correct vlan. 202:51966' - Wrong password. REGISTER 11 headers, 0 lines REGISTER sip:voip-co1. I have a number of Yealink phones that won’t register. I downloaded, compiled and installed asterisk without problem for use of sip phones only, without any Digium or telephone hardware at all, but my netcard. I have a sip Hey Community, I was wondering if I could get your help. 2. Perhaps run "pjsip reload". Status - The status of the registration request. The asterisk (at home) is connected with an adsl with dynamic public IP. ). Every time the device sends a registration, I get the following notice: Stack Exchange Network. 2:5070' Solved SIP register failed. 2. 190. The results of the test are directly related to the success/failure of the SIPp scenario - if Asterisk Value ISDN Cause codes (Q. 100 NAT IP for Asterisk Server 2: 200. I am aware that asterisk 16 is EOF but I just want to see if there is anything that caused this abnormal behavior? Relevant log output After a reload of Asterisk it registered again and all was fine. in CLI it says: [2014-11-10 16:33:05] NOTICE[12307]: Asterisk can quietly fail to open the http ports. so module. A minimal configuration consists of setting a 'server_uri'and a 'client_uri'. Does it resolve I am having trouble getting SIP phones to register with Asterisk. This tool is not related to fail2ban project. This is just a fancy way of saying he makes sure the ship is pointed in the right direction. 11 firmware. [2014-05-06 06:20:52] NOTICE[2107]: chan_sip. 16 623×977 146 KB. conf¶ [registration]: The configuration for outbound registration¶ Registration is COMPLETELY separate from the rest of 'pjsip. Viewed 1k times 0 . I will also type in The “header” endpoint identifier was extracted from the ip endpoint identifier by ASTERISK-27491 and will first be available in Asterisk 13. It requires Linux with iptables and ipset. I am attempting to use this as a local service acting as phones for an office like setting. conf ( obviously got overwritten ) Added below lines to pjsip. Have I picked a bad provider or is there a way to find out what the sip . For Asterisk to issue a new REGISTER request with Authentication I had to define realm specifically with the value we see in the 401's WWW-Authenticate header. Question: Is it possible to let Asterisk retry forever, even the “Failed to authenticate” occurs? Hi all, hoping for a pointer here as I have been looking at this for a few days and there is clearly something I either don’t know that is simple or something weird going on. 0) and two baresip clients (version 0. 2). I can capture it using wireshark/tcpdump, but the packet arrives at FreePBX and core-asterisk-running-2024-10-21T07-18-40+0200-info. txt ( 1) connection_schema_002. asterisk作为sip客户端，注册到其他sip服务器。2. Domain - The address portion of the registration. 0, Asterisk 1. 38 EC mode These are the phones I have registered – I do not understand ext 124 - but it is working just fine I cannot get Extension 111 to register - and, of course, it is the one I need most. 0 Via: SIP/2. yyy. Disallow use of extension outside the LAN is unchecked as required (remote site, connected via site to site VPN). conf and include these dialplans into your users context. I would then watch -n 1 "cat /var/log/asterisk/full" before you try registering a device again to see if you can see anything being logged for errors. c:25797 handle_request_register: Registration from '' failed for '192. I am trying to register a Grandstream HT802 for use with FreePBX 13. This topic was automatically closed 7 Trying to register a sip client to my asterisk server often (just about 90% of the times, not always, weirdly) results in 401 Unauthorized errors. Post by Olle E. 8. 1(1. 31:5060;branch=z9hG4bK027fe0e3 CSeq: 102 REGISTER User-Agent: Asterisk PBX I am trying to register twinkle with asterisk. 200 I am attempting to use Hylafax+IAXmodem on a dedicated server to send via an Elastix server on the same LAN, but I cannot seem to get the IAXmodem on the fax box to register with Asterisk on the Elastix box. Hi guys, I'm new to asterisk, but I'm given a try. You see on the log that asterisk try to register on "Jan 11 18:31:59 WARNING[1359]". system (system) Closed December 29, 2022, 3:10pm 8. Put information on the gui for phone activation. He originally started in the community submitting simple patches and grew into improving and creating new core components of Remote computer with static ip trying to register on my asterisk(1. Cause - What caused the rejection of the request, if available. it. May 25, 2022 #1 Hello, again I have problem with the telephone that shows SIP Register Failed. 3 into my Cisco 7942. 2 in Ubuntu server 16. My study so far tells me that REGISTER is only for asterisk to reach or forward the INVITES but not to authenticate an INVITE request. Here is the log: I'm having some issues with the registration of an extension configured in a Huawei HG8240H with the asterisk server. conf as opposed to fromuser=. 40 setup and running. If an exact match on both username and domain/realm fails, the match is retried with just the username. conf ( no avail ) [PjSipVoIPmsCG] max_retries=1000 forbidden_retry_interval=30 fatal_retry_interval=30; Here is my pjsip. The issue i'm facing is that the router is trying to register as username like NOTICE[4862]: chan_sip. Same can be if you use nat=yes and you have no nat. Ask Question Asked 10 years, 2 months ago. conf contains these lines: [guest] username=guest type=friend context=public callerid="Guest IAX User" secret=test auth=md5 iax2 show users indicates that Asterisk is aware of these settings. Arguments¶. The 'r' is re-attach to asterisk. * If Asterisk has stopped sending REGISTER requests, then either the maximum number of retries has been attempted or the response that Asterisk received from the registrar was considered to be a permanent failure. 6. So now that you are attached; I would say try reloading. NAT IP for Asterisk Server 1: 100. SYSTEM. 11. The asterisk logs do not show the extension trying to register. – BTR Naidu. 15. asterisk -rx "sip show users" Again, if you use nat=no and your device after nat, it will not work. 58. When I get to the Asterisk command line interface and type sip show registry I always get the same output, State = Request Sent. i have successfully installed Asterisk server but when i am registering user on it it shows following error: chan_sip. Modified 8 years, 10 months ago. 0 and 15. 9w次，点赞12次，收藏9次。nacos启动报错【导致注册失败】get service name from nacos server fail//或No service to register for nacos client// 或nacos NoSuchMethodError// 或nacos registry, qit-service-provider register failedNacosRegistration{nacosDiscoveryProperties=NacosDiscoveryProperti_nacos registry, Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: FPBX-2. If Fail2Ban is still a factor - I do not knoww how I can “Clear” out any old fragments of banned IPs it’s holding. org and post the link here. 16:11344' - Wrong password any help will be Great. Registration is OK. If you want to get Asterisk to start sending REGISTER requests again after making configuration adjustments, you can do so by Hello, I am very new to the freepbx platform. 12. From the original sip. * You must list at least one method that also matches for AORs or the registration will fail. 7. *In the failing config pjsip10. One of the most common issues in Asterisk SIP trunk troubleshooting is SIP registration failure. Class¶. 1 Like. This module allows 'res_pjsip' to register to other SIP servers. This should show the port is open. uqx tiyg druhn zwtt cnnwn vawj ajp edf yvydty jjye vjdvzw adr tsl wcunx gvm