Sip show registry ===== Connected to Asterisk 11. The SIP clients could NOT register, so they were offline but "sip show peers" stated that they were OK. A lot of SIP switching platforms allow you to view register status, but just keep in mind it doesn’t mean the device is still answerable at that address, only that it intended to be. SIP. core set debug On the SIP provider side, most Internet VoIP providers have realized that the client supplied IP address is likely to be wrong, and ignore it. It doesn't seem to be platform specific, for example I have seen 29xx routers where it appears sip show registry. Hmm sorry but I'm a bit new to asterisk. 9. xxxxxx. Joined Jan 6, 2012 Messages 147 Reaction score 5. conf changes on the fly you will probably want to reload the file and reset your registrations, the following command will accomplish that – As the name would suggest this will drop all your registrations, so will be service affecting. 80. by gmcust3 » Sun Jan 02, 2011 12:11 pm . Add SIP Trunk. Outbound proxy = sip. 24. Feb 13, 2018 #15 Both "sip show peers " and "sip show registry" shows the trunk registering on port 5080, but this is what I have been saying. If you have been making /etc/asterisk/sip. In SIP, this is not always the case. RFC 3261 SIP: Session Initiation Protocol June 2002 The first example shows the basic functions of SIP: location of an end point, signal of a desire to communicate, negotiation of session parameters to establish the session, and teardown of the session once established. show sip-ua statistics 4. flowroute. Each phone will have different number. voipfone. There are additional commands let you view SIP registration cache information, and to clear and audit information Hi I am using elastix 4. But then again maybe I am spacing it. 200 ServiidorSNJ SITIO 2: Tiene la Dirección Latest Vitalpbx onpremise. Description¶. como no se puede registrar el TRUNK no puedo hacer llamadas al exterior, ya no encuentro ninguna solucion para este problema. Current status is that it's not working but we can ping and traceroute successfully. sip show registry lists the peers that you have registered to, not the other way around. Time callcentric. antisip. ms port 5080 or not. But according to the book, Network Warriar 2nd Edition, in page number 560, they are mentioning about this command "show sip-ua register status" and it will display the SIP trunk lines too. Get Volume Pricing. Here is the output from that book. Further reading: OP configured 2 SIP extensions, but used only 1 SIP client (2000) to connect to asterisk. absent is (usually) because your registrations are no longer going through. Commands follow a general syntax of <module name> <action type> <parameters>. Sip show Registry. *****. However, I would like to know whether a specific user has registered SIP server or not in Once done, you can then check whether the registration has been successful by using the following command: sip show registry You could use this cmd : sip show peers to see all extensions and trunks setted into Asterisk, and sip show registry to see the registry accounts. 16. you can generally cause the sip phone to begin attempting to register by changing its sip proxy or server to the wrong address, and then back to the correct address 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 At that point, the Asterisk command sip show registry should print a line showing that you are registered, meaning your trunk is established. Type these cmd into asterisk Within this guide we will show you how to enable and make use of the debugging modes built into Asterisk, these can be used to capture the RAW SIP messages, and display to screen or log to a file on the server. I have register my SIP Account as follow: SIP-server User: XXXXXXXXXX Password: YYYYYYYYYY registrar: registrar I have restarted asterisk and reload sip details. ca:5060 N 416XXXXXXX 105 Registered Tue, 02 Jun 2015 Esta es la informacion que resulta del comando "sip show registry" odcsv01*CLI> sip show registry Host dnsmgr Username Refresh State Reg. localhost*CLI> sip show registry Host dnsmgr Username Refresh State Reg. Any and all non-support discussions. I would prefer to perform an "automated" SIP registration (via cron script). For example when I ran it I only had back to Feb 1, because that is the start of my log on that server. Basic example how to register Asterik SIP Trunk. About Register String: Username:Password@Domain; Submit and Apply, then your trunk will Sip show Registry. 0. The show sipd endpoint-ip command displays information regarding a specific endpoint, and the show registration command displays statistics for the SIP registration. [ 217 ] In SIP, this is not always the case. I have a fully functioning Asterisk box behind a nat router [root@HDVOIP10 asterisk]# asterisk -rx 'sip show registry' Host dnsmgr Username Refresh State Reg. These examples show the SIP details with call flows that include SIP User Agents and Clients, SIP Proxy and Redirect Servers. show sip-ua registration passthrough status 3. The configuration examples below are for just one SIP Trunk. R1-PBX#sho sip-ua register status Line peer expires(sec) registered P-Associated-URI >sip show registry xxx. In reality most deployments foresee a process called registration (method: REGISTER) which allows a Asterisk SIP Trunk Registration Example. 249:5060 SIP/2. Phones got disconnected and reconnected. It helps to see if Asterisk has successfully registered with a remote SIP provider. log. ) and their current statuses. The key snippet of the code is shown below with an explanation afterwards. I'm a totally newbie in VOIP and elastix. c:25797 handle_request_register: Registration from '' failed for '192. Keep in mind that is only going to show within the time frame the log has. Unlike chan_sip, it is not implemented in an obnoxious way. show sip-ua timers DETAILED STEPS Step 1 show sip service Use this command to display the status of SIP call service on a SIP gateway. d/vicidial file setting zap_mod=voicesync [root@HDVOIP10 asterisk]# asterisk -rx 'sip show registry' Host dnsmgr Username Refresh State Reg. show sip-ua register status 3. R1-PBX#sho sip-ua register status Line peer expires(sec) registered P-Associated-URI There are two ways to view basic SIP registration cache statistics. This article describes how you can check the extensions registered in the The UA doesn’t know anything has changed, so no REGISTER is sent to refresh, and messages from the SIP server are sent to the old address. 53. If you're embedding on your own page or on a site which permits script tags, you Solved: Hi, I've found that some CUBE instances don't show SIP registration messages and replies when you enable the normal "debug ccsip mess". However, you can also use the sip show peers Asterisk command to check your IP, as well as the port of your extensions. conf: Display specific SIP user: sip show registry: Display SIP registers: sip show settings: Display SIP settings: sip set debug on/off: Set SIP debugging: sip set debug ip [IP address] Set SIP debugging with filter by IP: sip set debug This guide introduces SIP registration with Yeastar VoIP PBX. . I have a 2811 with CME 4. so should show the following: Module Description Use Count Status Support Level res_pjsip_registrar. Note: All screenshots Click on Add Trunk and then Add SIP (chan_pjsip) Trunk: Step 3 - On the General tab, give the trunk a name and specify the outbound CID to use if a call makes it to here without an external CID. I then attempted to create a SIP gateway. Any ideas? Vieri I then made a call on the dialer to verify it worked correctly, which it did. Make an informed choice for your business communication needs. Registration expiry/Proposed expiry = 1 minute or 60 seconds. Show SIP registrations (text format). HTML: Markdown: Embed the player. Internet was down for about 2-3 hours. xx:5060' - Wrong password that's what I configure in sip. 2. This can be duplicated for any number of SIP Trunks. XXX:5060 N . Example: Sip show Registry. Sip reload asterisk1*CLI> sip reload Reloading SIP Under the PBX tab, click on PBX Configuration. Useful in places where scripts are not allowed (e. The following sample output shows that SIP call service is enabled: Router# show sip service SIP Service is up grep 'Registered SIP' /var/log/asterisk/messages That will go through your log and only show you all the lines where "Registered SIP" appeared in the log. 69. Enter the Register Name, User Name, and Password that SIP Registration is the process of binding an endpoint's AOR with its location. 20. For more examples of SIP call flows and best practices. sip show registry; Description: Displays a list of all external SIP registrations. 150 USER/ANR (none) , format: nothing , hold:no , SIPshowregistry¶ Synopsis¶. 144) and traceroute it. 7. " SIP Server/Proxy/Registrar = sip. Then, do the below command. com (216. How to Register. Test Numbers. SIP provider. Thanks, Wiley -----Original Message----- From: [EMAIL PROTECTED] For example, if your PBX has the IP address 192. Verify the state is Registered. Download ZIP Star (0) 0 You must be signed in to star a gist; What is SIP Registration? A registration is a connection between a device and a service provider's network. show sip-ua register status. 30. 202 D 5060 Unmonitored. Figure 1 shows a typical example of a SIP message exchange between two users, Alice and Bob. 0 everything works fine but cli shows unknown sip registrations with my current code running my cli output -- Hungup 'DAHDI/i1/9560790782-2fd2' [Jun 24 14:55:24] NOTICE[3 UserParameter=asterisk. The SIP Endpoint sends a SIP REGISTER request to a Registrar, containing its AOR, The example below shows how the To field of the child call leg is populated with the username of the SIP INVITEs To field of the parent call leg. Example: Device> enable Step2 show sip-ua registration passthrough status DisplaystheSIPuseragent(UA)registrationpass-throughstatusinformation. I am using asterisk 11. We need to call Croatia, and the provider gave us info that we should send the numbers in this format 385 xx xxxxxx. THen when I do a sip show registry I see "Request Sent". 101:5060 osaka 105 Registered Sun, 22 Apr 2007 19:13:20 I am pretty sure I could see all my registered phones yesterday via "sip show registry". SIP; using SIPSorcery. Some providers default to this, some you have to ask to turn that option on. I know that I need to register the SIP trunk under sip-ua config mode. Step 3: Edit extensions. Once 'sip show registry' showed up the trunk as registered however it didn't show up on web console as active registration. Explore the differences now. 115. torrent*CLI> sip show peers Name Hi, When I do a “SHOW SIP-UA REGISTER STATUS”, I got two records: one is showing ". The next step is to configure the phones themselves to communicate with Asterisk. Other settings should be left at default. Time 192. But we replace the analog circuits for a SIP trunk with the local provider. Created March 3, 2010 01:39. 33 D No No A 44973 OK (50 ms) ST SIP Registration Date[ दर्ता मिति ] * Payment Method [ भुक्तानीको किसिम ] * --Select Payment Method-- Connect IPS ESEWA FonePAY IMEpay Khalti Namaste Pay NIBL EBanking Prabhu Pay Thaili Digital Paisa SIP show peers SIP show registry ``` این دستورات وضعیت ترانک را بررسی کرده و نشان می‌دهد که آیا ترانک به درستی برقرار شده است یا خیر. XXX:5060 N +0XXXXXXXXXX 120 Request Sent This ensures that the SIP REGISTER is authorized, i. تنظیمات تماس خروجی. It means that the phone proved that it is supposed to be able to place and receive calls. And I couldn't make a correct dialplan entry, so if anyone can help I would be extremely thankful. It can be caused by: Incorrect credentials: Ensure that your SIP trunk When using chan_sip you can tell whether or not your phone has registered successfully to Asterisk by checking the output of the sip show peers command at the Asterisk CLI. Like with most concepts in PJSIP configuration, outbound registrations are confined to a Hi Nadeem, thanks for the detailed answer. If the Host Now I can ping sip. CSCuu97283; CVE ID "Compare peer SIP trunk vs register SIP trunk with us. Post by Wiley Siler All my SIP phones are still working and all my dialing is still working, so I did not think it relevent. 2 that is working fine for SCCP handsets, but I would like to use a couple of (non-Cisco) SIP devices. Time I also set sip set debug to 15 and sip set verbose to 15 but nothing! I checked sip-vicidial. When I ran the command sip show peers on asterisk CLI I was able to see which phones where connected and which phones where disconnected (unreachable). XXX:5060 N +0XXXXXXXXXX 120 Request Sent NECESITO AYUDA, TENGO PROBLEMAS CON CONECTAR DOS ELASTIX NO SE REGISTRAN YA LEI LOS MANUALES Y TEMAS DE OTROS CASOS Y NO LOGRO QUE SE REGISTREN, NO SE SI ME FALTA ALGO MAS POR HACER ANTECEDENTES ServidorP SITIO 1: tiene la Dirección 172. At the Zabbix Server a "Host Item" needs to be configured for the "Host" describing the Asterisk Server. Please see attached screenshot. If the registration fails, this command will highlight the reason. Trunk outgoing caller id format Trunks may be configured with a outgoing_caller_id_format attribute(see API reference ) which controls the type of formatting the trunk operator expects for the caller id number value. Time 173. XXX:5060 N +0XXXXXXXXXX 120 Request Sent rdegges / sip-show-registry. But "sip show peers" says the trunk is offline. We thank Jared Busch from Bundy and Associates for writing this guide. conf set the outbound CallerID name and append "000" as a prefix to all outbound calls. Disable SIP-Trunk and then enable: working. Moderators: gerski, enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, Michael_N 6 posts • Page 1 of 1. Under that mode, I need to config the registrar Hello, How looks username field in output of asterisk -rx "sip show registry"? limez17. Hi. XXX. These commands still remain. In your extensions. localhost*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description username/username 91. ; sip show registry Show status of hosts we register with ; sip set debug on Show all SIP messages ; sip reload Reload configuration file The sip trunk connection to the server is established, but this connection is not stable and is sometimes lost. net. Instead, they force the registration to use the peer IP address of the incoming connection. sip show registry; Shows the registration status of your SIP trunks. using System; using SIPSorcery. conf with outbound dialing modifications. c:15170 sip_reg_timeout: You can verify a successful registration with the use of the iax2 show registry and sip show registry commands at the Asterisk console. The status for all phones say "unkown" the last time this occurred, while the sip trunks were "registered". If your client can't receive calls and its status is OK, either the router it's behind has closed its NAT if your issue is with your sip phone registering to your vicidial server, you should be looking at "sip debug" to see if there is ANY activity being generated when the sip phone first attempts to register. tine Member. No such command 'sip show'. Background History. 0/UDP 10. 1 It shows the carrier listed in "sip show peers" but it doesn't show it in "sip show registry". If the endpoint is a SIP proxy service (as opposed to a user agent) , Asterisk will 1. With everything working as intended. One of the most common issues in Asterisk SIP trunk troubleshooting is SIP registration failure. Subject: Re: [Asterisk-Users] Sip show registry returning nothing On Fri, 11 Mar 2005 12:48:49 -0700. in a project's README file). If it fails then I can spawn a "rescue" script. برای تعریف تماس خروجی، به مسیر زیر *CLI> sip show peers Name/username Host Dyn Nat ACL Port Status toronto/osaka 192. -Removed the sangoma card from the dialer and put it in the gateway -Added Amfeltec PCIexpress timer to the dialer and modified /etc/init. sip_user[*],asterisk -r -x 'sip show registry' | grep $1 | cut -c 71-91. Connect to the asterisk console by running the following from the command line: asterisk -r Verify that Asterisk is registered to Callcentric with the console command sip show registry This is helpful for diagnosing issues with a specific SIP client. Use this command to check if your SIP trunks and extensions are properly registered and reachable. No outbound or inbound possible. 12:5061 as the SIP Registration Server for your UA. Why is it giving up? Is there a way to fix this with a setting? And since many versions of VitalPBX the STATE is not Do "sip show peers" and "sip show registry" on the Asterisk CLI to see whether you are connecting to voip. I got a problem of incoming calls can't hit the correct VOIP dial-peer, it I have used the command "sip show registry" and both trunks appear to be registered. 168. freepbx*CLI> help sip No such command 'sip'. dIŽ±L¹ 'Üâá= É 9c¦"© ¸ç†÷ðÞ H × É•† ÿ›J Š«ýŠŽ¤¨B ­CÌ­¤Ýý1–vï¦ Ö*è”åNîÖ çTºü¥Ë"Nï¢uUºuëÇ0«Î];ßþmb Ÿä1ÌÜíùno› £D 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 1 offline] So "sip show registry" says that registration is succesful. Is this possible? I can't find any good documentation on whether this is even possible, let alone how to do it! I am partway there, but when my SIP device tries to login i Type 'core show license' for details. I'm using Asterisk to register sip client but it shows me this message: NOTICE[3212]: chan_sip. A Registrar takes the info received in a SIP REGISTER messages and stores the IP Address and contact info in the form of an Address on Record (AoR). e only a valid device associated with the valid user should register. 0 Via: SIP/2. 164 numbers that a Session Initiation Protocol (SIP) gateway has registered with an external primary SIP registrar, use the show sip-ua register status command in privileged EXEC mode. How that can be? Would you advice where to look to fix this. Display specific SIP user: sip show registry: Display SIP registers: sip show settings: Display SIP settings: sip set debug on/off: Set SIP debugging: sip set debug ip [IP address] Set SIP debugging with filter by IP: sip set debug peer I have tried physical phones, softphones, IAX2 not even reach the server keeps registering and the asterisk logs shows nothing, but the SIP at least says wrong password, but the password authtenticating is correct, I have used the default, I have changed it, default and password shows correctly in phones table in asterisk database in mysql. conf but inside it I have only phones from [8002] to [8020] [/code] Registering Phones to Asterisk. تعریف تماس خروجی ۱. If you are using a third-party server, The display name is the account name shown on the device’s calling screen. The way we have configured the accounts in the SIP channel driver, Asterisk will expect the phones to register to it. Description¶ Lists all registration requests and status. use "sip show registry" inside of asterisk to display the ougoing registrations; enable sip debugging: "sip set debug on" (shows the sip traffic within asterisk cli) force a Show SIP registrations (text format). 5 on 2911 with 23 SIP phones (3905, 6961, 8941). Time 10. " while the peers up , even i can make internal calls! i need to know if its a configuration problem ? localhost*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 200/200 192. Surely, a "real" sip registration is more reliable then "sip show peers". Scenarios include SIP Registration and SIP session establishment. 12, you will need to use 192. Lists all registration requests and status. com:5060 N myusername 105 Registered Tue, 19 Mar 2013 04:59:47 My trunk config: username=xxxxxxxx type=peer secret=xxxxxxxxxxxx qualify=no port=5060 insecure=very host=sip. Like with chan_sip, Asterisk's PJSIP implementation allows for configuration of outbound registrations. 121. If the SIP line registers but the extension doesn't, is there an obvious reason for this? Please help! Thx!" asterisk-CLI> sip show registry Host dnsmgr Username Refresh State Reg. By: Olle Johansson (oej) 2006-11-14 06:03:25. Any other state indicates communications problem (firewall / NAT issue) between your Issabel server and GoTrunk network or incorrect VoipNow uses Kamailio for the extension registration process. show sip-ua status 5. *" which registered status shows no; another one is the actual SIP number which registered status shows yes. 37:5060;rport; Running module show like res_pjsip_registrar. provider. 178:5060 N dario 120 Request Sent 1 SIP registrations. 78. 2. [root@HDVOIP10 asterisk]# asterisk -rx 'sip show registry' Host dnsmgr Username Refresh State Reg. com context=from-trunk canreinvite=no nat=no my register string FreePBX Error No such command sip show | How To Solve No such command sip show (type 'core show help sip show' for other possible commands)https://technolog FreePBX 13+ with SIP Registration Getting Started. Looks like my extension to register even though the SIP provider number seems to register fine. In menu select, go to channel drivers option and check the checkbox chan_sip [Probably unchecked]. so PJSIP Registrar Support 0 Running core To register a SIP account, the following information should be obtained first. sip show registry Reload SIP configuration, and Re-register Clients. Zabbix Server. You can verify that your own registration was successful by running sip showregistry from the Asterisk console: *CLI> sip show registry Host Username Refresh State Reg. 130 Yes Yes 5060 UNREACHABLE SIP is a peer-to-peer protocol where the roles client – server and exchangeable depending on who starts a session. My approach is very similar to it (using asterisk -x "sip show registery", parsing the results, and sending an email when necessary) User If you’re already a Zentrunk customer, we have SIP registration instructions that show and tell you exactly how to register a SIP endpoint. Show Gist options. sip show registry lists registrations between SIP servers. Registrations will follow as separate events followed by a final event called I can check a user registration if I type show peer username on Asterisk CLI. I am on AAH BTW and info in AMP and in direct check of the confs checks out fine. For some reason the qualify=yes option was giving me a lot of problems. venturinog this is the result: [root@HDVOIP10 asterisk]# asterisk -rx 'sip show registry' Host dnsmgr Username Refresh State Reg. Why your netgear GS Additional comment actions. A few weeks ago, we have the PBX connected to the local telephoy system over a four analog voice circuits and have the SIP trunk for international calls configured, registered and working. I type the command "sip show channel" in asterisk cli, it shows the following result peer:192. show sip-ua register status [secondary] I waited 1 minute but sip show registry shows me Code: Select all go*CLI> sip show registry Host Username Refresh State Reg. Time sip. Registrations renew periodically throughout the day to prevent fraudulent calls. The outbound "From:" section of an outbound SIP Invite request should look like this: اما در ابتدا بگذاریم که در Sip Show Peers بررسی کنیم و ببینیم آیا این Trunk ما از طریق پورت ۵۰. freepbx*CLI> help iax iax2 provision Provision an IAX device iax2 show registry Display IAX registration status iax2 show stats Display IAX statistics iax2 show threads Display IAX helper thread info Name: it is the name which will appear in the outcall interconnections list, Interface: this is the channel name (for DAHDI see DAHDI interconnections) Interface suffix (optional) : a suffix added after the dialed number sip show The cause 20 sub. 000-0600 Please follow the bug guidelines and upload a SIP debug of the whole transactions - see serge-v's note earlier. Thousands of businesses in more than 220 countries trust Plivo’s cloud communications platform. When I do Sip show Registry , It shows me Auth. Registration is simply a mechanism where a phone communicates "Hey, I'm Bob's phone here's my username and password. xx. But next time we restarted asterisk the registration kept on timing out. 0~dfsg-1ubuntu1 currently running on torrent (pid = 518) torrent*CLI> torrent*CLI> sip show registry Host dnsmgr Username Refresh State Reg. Obviously, 2 SIP clients must be connected to asterisk before the call can be routed between them. After 5 hours realized: Trunk said “registered” but it was not. sip show peers; Lists all SIP peers (trunks, extensions, etc. but cisco don't reply on this and strangly don't show it in sip debugs. 4. The UserAgentRegister contains an example of how to register a SIP account with a SIP Registrar. For example: sip show peers - returns a list of chan_sip loaded peers; voicemail show users - returns a list of app_voicemail loaded users; core set debug 5 - sets the core debug to level 5 verbosity. T. I currently check the return of the "sip show registry" command, and parse the results. Thanks to that option I was able to tell if phones (peers) where connected or not. sip reload sip show registry I got following: NOTICE[9143]: chan_sip. com fromuser=xxxxxxx fromdomain=siptraffic. net My main problem is, after much reading and tweaking, I still cannot get my CUBE gateway to register with my sip provider. 0, i have extension 200 and trunk ST , sip registry gave "0 SIP registrations. Copy code block. siptraffic. show sip service 2. App; Sip show Registry. Configure outgoing and incoming settings. Overview¶. Understanding SIP registration basic; Troubleshoot extension registration issues; SIP Extension Registration. I try to register with x-lite (soft client) with same parameters and it's worked, and in capture i see that x-lite send same register packet as cisco, then recieve same 'unauthorized' message from provider and then again send register but with authentication and 'nounce I'm preparing CME 10. show sip-ua registration passthrough status detail DETAILEDSTEPS Procedure Step1 enable EnablesprivilegedEXECmode. com:5060 N 17772807810 45 Registered Thu, 10 Sep 2015 11:00:55 1 SIP registrations. It £¤˜ QYü! ‘šõC€FÊÂùûGèð9ï?Ûkñ¿Nª²ØŸKn&ÉÄ€ø cÇž! wÇù9éït¹ ° !1’°q{\u û·XìÞü7gÿkÿ9Yìo˜œ#Ì T{t>‚ #––ò* ^C ] üÚÛÕÕæÿ½¯Ú;óÛ ‹ò;‡. If an extension is listed, then it is either registered or set up with a static IP port. ۶۰ مرکز مقابل را می البته اگر ترانک شما از نوع رجیستری باشد، باید دستور sip show registry را برای ترانک شما So if we want to send a SIP message to Bob, we look up Bob’s IP in our Address on Record list, and send it to that IP. To display the status of E. No official PJSIP Support yet. g. If there is no UserID association to the device then registration rejects with 401 response code. Registrations will follow as separate events followed by a final event called 'RegistrationsComplete'. SIP Server/Proxy/Registrar Port = 5060. I have configured one trunk at OpenIP but when I do sip show peers in the CLI, I see my trunk unmonitored but not OK My trunk is: type=peer host=Sip3. com:5060 N username 120 Request Sent 1 SIP registrations. REGISTER sip:10. root@incrediblepbx:~# asterisk -rvvv # then do incrediblepbx*CLI>sip show registry incrediblepbx*CLI>pjsip show registrations # There is help available incrediblepbx*CLI>core show help incrediblepbx*CLI>core show help sip incrediblepbx*CLI>core show help pjsip incrediblepbx*CLI>core show help [topic] Show / Hide Table of Contents. Sent. Try checking it with "sip show registry", probably says something like "auth sent". 48. It looks like I need to put something in [outgoing] so that when any outgoing calls have finished the dial status and trunk name will be put in a file somewhere to be viewed or if dial status is chanunavail something happens which can trigger a script which I'll make later. pjsip Use snippets below to display a screenshot linking to this recording. Registration associates user’s identification, or Command Syntax and Availability¶. Type the following command: sip show registry; Click Execute button. In my case i have installed asterisk You can verify a successful registration with the use of the iax2 show registry and sip show registry commands at the Asterisk console. 1. cvqfa rmcj miwkf zte xcrxen drnu fcea pzxyfpj qchyp muifdw