Freepbx disable pjsip. Had to reboot to get them back online.

Freepbx disable pjsip 14 with Asterisk 13. c: Contact 3210/sip:[email protected]:5060 is now Reachable. The problem arises from the fact that SIP Server is FQDN, but REGISTER goes to wrong IP. aor. Hi to all, In a scenario with FPBX, users send 183 with the required 100rel so parties expect to receive the PRACK and then respond OK, but Asterisk just responds OK to caller PRACK and does not forward PRACK to the callee then callee resends 183. What I have found is on the old system, a standard hello everyone, i’m kinda new to this so I explain my problem I have a freepbx central in operation which has a main IP and everything works correctly, my service provider gave me a SIP line to configure said trunk, perfect previously, I have already configured trunks in the cloud and I have not had any problem , My provider gives me the trunk by Ethernet cable with FreePBX does set remove_existing=yes on the PJSIP AOR’s it creates. However, now the PJSip tab is missing from SIP Settings and the Asterisk Info report no longer has Chan_PJSip Info listed there. billsimon (Bill Simon) chan_sip is already disabled by default in FreePBX 16. Their TELCO insist that we have to adjust maxptime value to 20 or to a multitude of 20, instead of the default 150. 66-19 Advanced Settings “Disable Follow Me Upon Creation” (True) enables new PJSIP extensions Follow Me anyway. I tested this by unregistering one of the units, As the title mentions, I’m sharing what I came up with to solution for an instance in which I needed SIP and PJSIP to message each other. In reviewing security settings it was suggested to disable SIP Credentials if using a static IP host, when I did that we lost outbound calling. xxx. Then, when attempting an inbound call, audio passed for just a second or two and the call was dropped. Freshly installed FreePBX Distro (FreePBX 15. I copied every single configuration detail from the older PBX system to the newer one. ” No. All option packets are getting dropped in this case. Does FreePBX have any settings for PJSIP’s jitter buffer? I can see that it’s implemented in the documentation for PJSIP, but I can’t find any way to enable it for PJSIP in FreePBX, just SIP. 0, Firmware 10. 24. Elide that option from your current operation. 447 msec to the FreePBX. Set all three Retry Intervals to e. 20 with Asterisk 13. 168. 7. Registration, incoming and outgoing calls work fine as long is not set to failover to another trunk, just the trunk constantly shows offline. Hi all, My SIP provider requires using DNS I’m trying to set up a PJSIP trunk between two FreePBX servers, but I’m not having much luck. This is a very low traffic system so I can’t be sure of more exact timing but the last Do the results of sip set debug on or pjsip set logger on output results to a file? SIP set debug. 5 Grandstream GXP2000 I have two inbound routes and two extensions (200, 400) supporting two phone numbers. reboot : Request phone to reboot. Enabling them for SIP is a I don’t think I can, I am not a big supporter of chan-pjsip for trunking , and good old chan-sip is well documented I had the same bahaviour on my 15 box, distro latest version, pjsip extensions, after applying config on a bunch of module updates. Just disable chan_sip altogether (in Advanced Settings) and set PJSIP to 5060 (Asterisk SIP Settings). This will include Direct Calls, Ring Group Calls, Queue Based Calls, IVR transferred calls, etc. Use the default ports until you really understand how ports work. ms, and I followed this documentation: SMS-MMS :: VoIP. 1. FreePBX can be installed manually or as part of the pre-configured FreePBX Distro that includes the system OS, Asterisk, FreePBX Use only PJSIP and disable chan_sip (in Advanced Settings - SIP channel driver). I . PJSIP trunk configurations are filtering video CODECs ( H264 , VP8 etc) outbound. It just doesn’t accept any calls. 0/24 network. I copied the addresses from a notepad Hello, I am currently working to migrate our production server to a server that is running a newer firmware. Immediately after updating the Firewall module, the I have (as per previous thread of yesterday!) decided the best course of action is to disable EPM and go back to DPMA. 2, all modules upgraded) I’ve created two chan_pjsip Extensions, 11 and 17, they can call each other but when i hangup the call, the channel won’t be closed. I´ve managed to configure one pjsip trunk in FreePBX (15. FreePBX binds Chan_PJSIP to 5060 for UDP and 5061 for TLS. But I am also using chan_pjsip. I have jitter and dropped calls occasionally and after lots of reading I think I need to set sip set debug on and pjsip set logger on. This is the solution: Indeed, the option “Enable CDR Logging” must be set to “Yes”. Responsive Now need to move a SIP trunk over to PJSIP but very noisy logs with warnings and errors. However, I did not manage find it in a “new” chan_pjsip driver. So is there any specific config in pjsip for that? any idea? Current call flow is: A–INVITE—> FPBX Remove registration on the pjsip trunk (set Registration to None or delete the trunk altogether). For more information about the configuration of FreePBX, please see the FreePBX wiki. I’m configuring an goip9 gsm-to-sip gateway. Question: I can’t find the place to set that in web front end. kbaker521 (Keaton Baker) I know that I can disable the warning, but I wonder if I could sign it for my use, I would be warned if someone else changed the code without my knowledge. Turn that off RIGHT NOW. Hi all, With an older FreePBX Distro pbx (10. ms Wiki SIP/SMS with Howdy, w. I can see INVITE packets being sent where the phone attempting to initiate the video call sends the video port (e. Registration goes to PBX but Asterisk is using the private IP of the device and not the public IP. But they don’t. Please advise. I have a laptop with softphone on a 192. Submit, Apply Config, restart Asterisk, test. We also only have outbound information configured in the trunk settings, the inbound information is left empty. This is not SIP, it’s networking. The ca. Also I This is not a Chan_PJSIP related setting. c:1012 registrar_on_rx_request: AOR ‘goip1’ has no configured max_contacts. If you use certificates that are out of date (before or after/expired), things are a no go. First, we tried using the setting taskprocessor_overload_trigger=none (this is available in FreePBX 15+ GUI under Settings > SIP Settings) which should tell Asterisk not respond with 503 ever but that this is my first pjsip installation, and set up my extensions as pjsip (big mistake), making the assumption I had to because of port 5060. 2 was released in 2005. But when you reject a call on one of them, then the other endpoints keep ringing. conf and put in this: [default-yealink] Content-Type=>message/sipfrag Event=>ACTION-URI Content=>key=Reset Save -> Reload. Both are configured essentially identically, but only one can receive calls. SANGOMA PHONES ONLY rebuild : Rebuild the config files for an extension/template/all rebuildupdate : Rebuild the config files for an extension/template/all and tell phone(s) to check config. 4 I ran tcpdump and get 10. Endpoints. In my FreePBX, I tested removing fromuser which did the trick and now I can see the DID coming on the external phone; however, outbound calls are till dropped once reaching 20s exactly. By default, 5060 and 5160 are the ports used. Once all trunks are successfully registered (I have 5 PJSIP and 5 SIP), ever 2 or 3 days or so, one or more of the PJSIP start showing the status “Rejected”. After clearing the PJSIP tab, applying and running the fwconsole restart command, the problem continues and the pjsip. If no luck, post details including the contents of pjsip. a single setting that is applied to all the endpoints. What does the Asterisk CLI show for pjsip show transport 0. Hi, I am using both sip and pjsip extensions on my Asterisk setup. All my PJSIP has direct_media enabled by default. remove : Remove an extension. r. Removed contact 'sip:[email protected]:60524;rinstance=be1ed6cd533e23f0' from AOR '100' due to remove existing 1412 [2023-02-06 10:37:58] VERBOSE[33214] res_pjsip FreePBX 13. But sometimes FreePBX is disabling my pjsip modules at startup by modifying the modules. Came into the office today, no changes made to phones or Pbx but the two extensions won’t ring with incoming call. New v15 distro with v14 restore. 66), incoming calls were always audio-only. I noticed in the log that the system is using a pjsip method for the chan_sip device. There are some extensions that has one way audio after 1 second of the call, I mean, when you answer the call, you can hear the first second of the call, then it goes to only one way audio. If you must use chan_sip, set registertimeout to 180 and registerattempts Hello, community. The server is in a DC on a dedicated DMZ and a valid IPv4 address. Just switched to Asterisk 13 and trying to test out pjsip. I tried converting my extension to chan_pjsip but now the phone cannot connect. c: AOR ‘’ not found for endpoint ‘Grandstream’ (10. Scroll down and you should see ‘Port to Listen On’ in the 0. Is there another setting that is overriding this? Only a small percentage of extensions want Follow Me configured initially. (SIP/2. 100. At first despite reloading Asterisk the Digium phones themselves weren’t finding an mDNS broadcast from the FreePBX server, but a full hardware This is not implemented for chan_pjsip, and calling that will do nothing. I didn’t see anywhere to disable PJSIP so if it is sending both, not sure how to stop that. In the above scenario, SIP signalling OPTIONS is affected by the qualify setting, which is present for both SIP channel drivers. If not, post details. My provider told me to register every 1800 seconds / 30min to remove Has anyone else found recently that PJSIP has become disabled “all by itself”? I’ve got a system here running the most recent FreePBX distro, and sometime between Nov 25th and today the “SIP channel driver” setting in Advanced settings changed itself from “both” to “chan_sip”. The issue is a lack of audio on PJSIP extensions on internal calls when connected from some public IP addresses. FreePBX 15. use of self-signed certificates on a Yealink T-5xI’ve a T-58V with firmware 58. In my case I needed to use the trunk elsewhere, so I disabled it and unregistered from the command line. 24 / Asterisk 16. As shown in picture, changing NAT = yes and IP Configuration to static in Settings > SIP Settings > Chan SIP Settings solved the issue for chain_sip extensions. 26 with Asterisk 13. 1 and ext. The trunk is working, i can dial in and out without issue so far. It also doesn’t have global settings, i. However, it’s still asking for my password. 29. The PBX has a public IP address and is one of many within the same data center and is the only system having an Hi, My trunk-provider requires P-Asserted-Identity to be set on outgoing calls to one of the numbers that we have with them to bill the correct number regardless of outgoing caller-id. Went back to a week in VM snapshots - same issue, rebooted network, temporally modified firewall to allow SIP connections from all external IP’s, You can create a trunk using either library. 729 yet, FWIW), but on-topic for this thread, it would still be nice to know how to turn the jitter buffer on FreePBX has a place for the force_report entry in the PJSIP trunk configuration. I ran the module admin from the CLI and fwconsole reload and that problem went away. Also kind of weird that even though its not implemented the documentation on asterisk, includes the params - it can be found at wiki dot asterisk dot org in the Asterisk 18 section Asterisk 18 Configuration_res_pjsip. 23 I installed FreePBX on a VmWare server and registered our two Lifesize Icon 600 video conferencing units to it. If you leave out the secret for an extension you will get a warning that it is bad practice to have a blank secret, if you click OK it will save the extension without a secret This is definitely a bug in Asterisk core, and in FreePBX core. I mean when I try to register a new ISP SIP trunk in a fresh Freepbx (PJSIP on port 5060) I can’t get it work (no matching endpoint found) due to port 5061 instead of 5060 an vice For the extensions in question, turn off Rewrite Contact and Force rport. What do you think might be going on? A user enables call forwarding on the My setups are usually latest Debian with a stable version of Asterisk and Freepbx. I changed all of my extensions back to SIP_CHAN, and modified the default port on the phones to 5160, and like magic the issue disappeared! Seems that the older phones just don’t work with pjsip. 0 401 Unauthorized) My FreePBX server: 10. i want to use the number of the incoming call We are currently going to upgrade our FreePBX server to the latest version. g: 216,221,223) and 3 local extension (e. 2) On ext. The other (200) fails immediately with ERROR[14481] res_pjsip. Turn off “Anonymous Inbound Calls”. This works when a call has one of the called ids from this provider as it gets set to the If FreePBX v17 is still a year out from being released, I’d say remove Chan_SIP from FreePBX v17. The system seems to be up to date. 12386) and hey David. 180 seconds. Set up your Outbound Route(s) and confirm that you can make calls. I have FreePBX 15 installed on a little Pi4 and can easily SSH into the unit. 0 installed. The issue is when I build the extension using the Extensions Module within freepbx I cannot get any phone to register with that extension. since I'm not able to organically reproduce the bug, to test it you can disable pjsip by hand: From FreePBX interface, open "Settings" > "Advanced Settings" find "SIP Channel Driver" variable and set it to "chan_sip" Submit I was moving the sip trunk to pjsip (with flowroute) but wanting to keep the endpoints on sip. For a pjsip trunk, on the Advanced tab, set Expiration as desired. Brand new SNG7 1910 setup with all current updates as of today, including to Asterisk (13. c: Request ‘INVITE’ from 'failed for ‘54. Initially I seemed to get on well; I created a PJSIP extension When first installed the PBX about 8 months ago, default was PJSIP and I flipped it to CHANSIP, everyting was then created as CHANSIP ( Trunks & Extensions ) CHANSIP was changed to 5060 and PJSIP to 5160. I do plan to upgrade to latest version some time in the next few months. Any response should do. So I have configured Disable source port rewriting; Set Conservative state table optimization; 1 Like. Their instructions are as follows for Outbound SIP Trunk settings; Host=THEIR IP Hello, I am testing freepbx on the german research network with German Telecom as provider for SIP-trunks. I have check RTP debug and I can see that when the call is answered, there are some When creating a pjsip trunk, it does not include the max_contacts value in pjsip. 8 KB (Obviously, replace the “Outbound CallerID” with your DID Number. In the freePBX pjsip extension settings I set “transport” to 0. This should be done by manually editing the config files, since FreePBX won’t allow to enter an empty password from the GUI. 4 No. It is very random. type=user is unlikely to work, as it would require the ITSP to set the the user part of I have recently installed FreePBX (version 14. So it would need to be turned off to stop using it. For example: [150](+) remove_existing=no So now coming back to a phone with max contacts set to 1. Providers. I have 4 remote (e. I have set the driver in advanced settings to just chan_sip, but if I look in my logs all I see is a lot of errors about pjsip (why is it listening?), so how do I get the chan_sip to work? Its compiled into asterisk, the . Now send a sip notify. I have a Distro install using Asterisk 12 and FreePBX 12. However, some people wish to use PJSIP for one reason or another. conf:rtp_timeout_hold=300 Changing to: Hi, I’m trying to connect a Lancom Router with a central FreePBX15 System. res_pjsip/pjsip_distributor. Trunks, chan_pjsip First, you need to create a FreePBX Trunk for your Digium SIP Trunking account. 0 (udp) section. I converted a few test extensions to PJSIP to hammer out all the kinks with the new driver. so file is there and it seems to be loading, but pjsip still FreePBX is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server. It started on 5/21/24 with no known changes to the PBX or ISP. The snom screens show DND is turned off. 15 FreePBX I have been fighting with one way audio for awhile now. 2) PBX has public IP and is NOT behind NAT. You need to set up your inbound calling correctly - do not use Anonymous Inbound SIP under any When using pjsip for a sip trunk I am having a issue with a pjsip trunk constantly showing offline due to the provider filtering on the “From” header in the options packets. The Option --detail = All mapping information. PJSIP does not have this option because PJSIP doesn’t have a concept for anonymous. Most of our phones are Digium D40s and are configured via DPMA. Anyway, I did return all configurations to the proper settings, I have internet now, I did more specific routes, PJSIP is now on port 5060, disabled chan_sip FPBX loses registration status for both PJSIP and SIP trunks overtime. 19, Asterisk 14. 0:5160 I am trying to add a new route from and I need to disable the registration. Right now, my extensions are PJSIP, if that makes a difference. Regards. But, it doesn’t matter what you fill-in there, because that entry doesn’t get populated in the . Does anyone have a barebones config for a working trunk they could share? In the Asterisk SIP Settings - pjsip tab, select the certs for use with TLS transport, and disable client and server certificate verification (because you are using self I have had a working system for five years, a recent update has changed how freepbx/asterisk is working with VPN connections I have: Office-A --> openvpn --> Office-B 10. But I cannot get it to apply. x. trixie_no5 (Gunter Treichel) August 21, 2019, 5:48am 1. What is weird is that Asterisk is Hi, I’m trying to make my Freepbx work with my provider. 22. There’s no built in mechanism in either chan_sip or chan_pjsip to disable the setting of maxptime. I simply want to know how I can make a video call with FreePBX. Since there is nothing in the extensions settings to disable or enable this, it would just be “on” because that is the default setting for it. In my test i called my PJSIP extension from cell phone, then i made blind Transfer to an internal extension. This might be helpful. The default behavior in FreePBX is when max_contacts for a PJSIP endpoint is set greater than 1, remove_existing is set to no. Your help will be very much appreciated, Thanks guys. I have two local phones (ext. Pentium5 (Oleg) September 10, 2017, 7:59pm 1. I’ve UNINSTALLED End Point Manager using Module Admin and then reinstalled “Digium Phones Config”. I’m not sure about chan_pjsip, my freepbx system is ver very old and probaby a legacy version. By default pjsip extensions are configured with directmedia=yes. I can create and register pjsip devices but not chan_sip. The Router registers correctly at the central system. 19 PBX Distro:12. it seems to be causing issues so i would like to disable this. s&hellip; Disabling PJSIP and Changing default FreePBX SIP port and enabling NAT support. 1 forwarding is activated to an external number Ext. People are constantly transitioning from office to WFH and call forwarding is constantly changing. c: Added Hi, There area few sip phone in our environment for which I’d like to disable all kind of inbound caller id display. conf:rtp_timeout=30 pjsip. Availability; Enabling IPv6 support in application using PJSUA-LIB; NAT64; References; Getting around blocked, filtered, or mangled VoIP network. For a chan_sip trunk, you leave Register String blank. 51. it is adding the following lines: noload = chan_pjsip. Looking at the pjsip logs, I could see these values: Contact:Contact: sip:w. I changed pjsip to sip but this is too simple to work. This is an issue that I’ve been fighting with for a while, since pjsip came along, since I really want to use it for handsets (because of the multiple simultaneous registration thing). But Asterisk rejects the registration with the error: WARNING[31744]: res_pjsip_registrar. I’m not sure why the FreePBX developers thought that you’d only I do not see how that is possible in the current freepbx pjsip trunk settings. ms trunk. You either stay with PJSIP for both FXS and FXO and disable all authentication for the FXO port or move to CHAN_SIP for the FXO port Keep in mind, you’re asking for help with custom configuration of PJSIP in FreePBX. Then when you are in less of a hurry learn about ports. 240 context=from-internal FreePBX PJSIP setup. FreePBX. Follow Me section settings: Yes, Yes, 23, 7, Taken at face value, the 488 means that the list of ciphers enabled in pjsip does not include the AES_CM_128_HMAC_SHA1_80 or AES_CM_128_HMAC_SHA1_32 offered by the phone. Do the results of sip set debug on or pjsip set logger on output results to a file? and using arrow keys and space bar, disable display of all packet types that How to setup PJSIP settings in Freepbx without authorization. mickecarlsson September 6, 2010, 6:25pm 3. I would expect Method not Allowed, to be considered a good response, as the intent of OPTIONS is to obtain a response without changing the state of the remote party. I am unable to find this option for chan_pjsip in freepbx. No. crt file should be loaded into the Yealink’s list of Trusted Certificates. asterisk. Create a PJSIP trunk in FreePBX: A. Something "weird’ happened to my installation where I had a “Reload failed because retrieve_conf encountered an error: 1” message. c: Added contact Each of these is configured using the Admin Web tool provided by FreePBX. The reason you are asking for this help is because you want to make GVSIP work. com This may seem like a trivial task but for me who knows little of VOIP/SIP its a major undertaking. I have sip on 5060, tls on 5061 and pjsip on 5160. If you were not using GVSIP, you would not require this help. The system is v15 with latest patches as of today. Running FreePBX 6. If I had their settings for a SIP trunk, I Hello Everyone, I’m very pleased to introduce some major changes to the Firewall module and how security settings are arranged in FreePBX. I found two default settings in the pjsip trunk that I think would be @sorvani: I am under chan_sip and not PJSIP unfortunately. I have another trunk set as a SIP and it works OK. No problem, so I went to set up PJ-SIP using IPv6. suggestions (I haven’t tried G. Logging into the Asterisk CLI and doing "pjsip show endpoint " may yield a bit of insight, otherwise someone from the FreePBX side may need to chime in. “Danger, Will Robinson. 1. If I try to change it to port 5060 it reports a conflict with SIP Port. I worked around that by adding it to the custom file. Can anyone tell me where to look please? I’m sure it’s obvious Solution: Enabling SIP Credentials at Flowroute allowed outbound calling, no more ACB message. g. It correlates. Disable PJSIP, change the chan_sip port back to 5060 (like it was on your old Hi. I am trying to use the old sip_driver chan_sip. c: Unable to retrieve PJSIP transport ‘udp,tcp,ws,wss’ This is the only log message when the call fails. Normally, FreePBX does not restrict the list. I have a SIM card that permits only 2 hours of free outgoing calls, then after this 2 hours, calls are extra invoiced. 1805 (Core) 14. We recently started seeing large chunks of phones (35-40 phones at a time) go unavailable. The challenge is understanding how to do this. 2 -> 1. XXX My Remote Extension’s IP: 86. is there any way in freepbx to hide all incoming caller id for a few specific extension? Thanks in advance. I currently have a live FPBX 2. I suspect this is an incompatibility with pjsi That will disable this message. Using Port Forwarding pjsua can also be configured in port forwarding environment, for both SIP UDP/TCP and media (RTP) transports. Extension is behind NAT. Hello, How to change/set SIP Registration Timeout only for 1 trunk? Stewart1 (Stewart) August 2, 2021, 4:02pm 2. After building a new PBX (SNG7-FPBX-64bit-1712-2 I think; not an upgrade or migration), incoming calls now default to video mode (just pick up the handset) displaying an unfriendly black screen to the Hi I’ve FreePBX 15 with Asterisk 16. 1 Like. Can an instance of free pbx have both pjsip and chan_sip devices? I did an install of freepbx on ubunut 14. 192. I would like to know if there is a possibility to deactivate a trunk or an outgoing route after a counter of time of outgoing call, and enable it again after a specific reccuring date. This did not help. Assuming that the macro will prevent this path being taken, it looks like most of what forwarding progress means is related to DAHDI channels, with Dial itself generating the tone signals, at a [119] type=aor max_contacts=1 remove_existing=yes maximum_expiration=2592000 minimum_expiration=1728000 qualify_frequency=60 default_expiration=86400 How can I set this correctly so it stays? I would assume that I need to use pjsip. XXX What I checked: -Router’s Port forward (5060 UDP and the API: pjsua_handle_ip_change() pjsua_handle_ip_change() flow; Notes and limitations; IP change scenarios; IP address change detection; IPv6 and NAT64 support. XXX. z:5160 c=IN IP4 outside. Restart Asterisk, confirm that the chan_sip trunk is registered and try an incoming call. We are in the test phase. I installed FreePBX 14. While all internally, local phones work, phones that are remote and outside of the local network aren’t transmitting audio to and from. 04. Inbound Calls are set to go to context from-internal. I Disable the video support Please help. 0/24 network I have I firewall forwarding from an external ip of say 1. Messages will fail between technology types without a way to distinguish which technology type asterisk should Hi guys, I have discovered by accident that I no longer can use PJSIP trunk as an outbound! This trunk is at Telnyx. 17. Then go into Admin -> Config Edit -> sip_notify_custom. FreePBX is licensed under the GNU General Public License (GPL), an open source license. 25) in an attempt to migrate away from using Asterisk 13 with config files only. Is SRV lookup supported by chan_pjsip? If “yes”, how can I enable or disable it? FreePBX. 11 system with the same phones and similar config where this problem doesn’t exist. NooNoo: It would be nice to have a possibility to add “other pjsip settings” in the The 401 is definately being generated by freepbx. Let me clarify. 16. I initially enabled the responsive firewall and configured the Cisco firewall to allow all traffic to the FreePBX host. Run away. You can disable VAD in pjsua by using --no-vad option from the command line. Appreciate any help. RTT: 32. We have a client with Freepbx 14. c: Request from ‘sip:7103@xxx. I get errors when trying to connect via IPv6 using extensions set up with CHAN-SIP as it says the address family is not supported. I have a PBX on a 10. i understand that the providers (voip providers ) are to handle the stir shaken and that there is no support to my askersisk setups from what the Providers tell me. Doing so would require code modification to remove it. Had to reboot to get them back online. my use case: User is at home, registered with a softphone (contact 1) they undock their laptop, and it registers on their wifi (contact 2) they come in to the office, laptop connects to wifi (contact 3) and they dock their laptop at their desk (Contact 4) so now, I have Yes, I know Chan_SIP is going away, but for the time being, does FreePBX 16 still have the Legacy Tab for Chan_SIP? FreePBX Community Forums FreePBX 16 Legacy CHAN_SIP option? PJSIP, itself, is a SIP driver that has been around since 2005. I would like to be able to stop using the SIM card (GSM This is really an Asterisk issue, but because this first appeared in these forums recently I’m posting this here. One or more of the SIP ones show the status “Request Sent”. conf files. If it fails, report what, if anything, appears in the Asterisk log. I’ve tried a grand steam phone and a couple of soft phones. sng7 Asterisk Version:16. conf looks like this [0. 30 shouldn’t the Extensions settings > Advanced > Call Waiting Tone = Disabled stop the call waiting beep on the extention when a second call comes in? I’m pretty sure it used to but doesn’t anymore. JessicaRabbit April 1, 2020, 12:04am 1. I set this up with PJSIP and have “Send RPID/PAI” set to “Send P-Asserted-Identity header”. On a pjsip trunk, you set Registration to None. Call comes in, pick up the handset, and you were on an audio-only call. 3. 2 questions: Is there any way to disable that behavior? (in the trunk configuration?) Is this really needed? 🙂 Thanks for your help! FreePBX. It only applies to Chan_SIP which has the allowguest= option. My current SIP trunk is through VoIP. conf using the Config Edit module in FreePBX. This issue has become more pressing to resolve since COVID-19 and WFH. Disable that setting. A packet Hi everybody, I have a problem using FreePBX 14 with Asterisk 15, new installation. That’s ok, and when you answer the call via one of them, then the others stop ringing- that’s ok as well. Command Options: fwconsole convert2pjsip [-a|–all] [-r|–range RANGE] To convert all chan_sip extensions To check your pjsip port, you can go to Settings → Asterisk SIP Settings → pjsip settings tab. The port I set to 5062, but the Cisco doesnt care and For several months I have had issues with disabling call forwarding. Upgraded to FreePBX 16 - no change. Double-checking the output of the Asterisk command line when firing the “cdr show status” command when the option is set to “No”: Call Detail Record (CDR) settings Logging: Disabled Mode: Simple And when the option is set to “Yes”: Call Detail Record (CDR) settings Logging: Its curious that even though its not implemented that doing pjsip show endpoint <id> shows the codec_prefs_* parameters for all endpoints though. 44 with Asterisk 16. It is random though. 18. 9. MAA August 2, 2021, 4:07pm 3. It tries to register to freepbx. crt file and a number of other files. Not an Asterisk bug, FreePBX sure. ) You also wouldn’t want to disable sip credentials in Flowroute. 65-29. ) (shown on the next screenshot with a red arrow) and Hi, I have a SIP Trunk with ITSP (OTENET from Greece) where I have configured a CHAN_PJSIP correctly. 65 64bit installed with Asterisk 13. Set general settings. 0 First of all, thanks for the hack. If I then re-enable the trunk, the registration of the SIP extension on FXS breaks and the FreePBX log fills up with “WARNING[2615] res_pjsip_registrar. There are an abundance of tutorials online for enabling SIP messaging for either SIP or for PJSIP, but they don’t intermix. aor_custom_post. Trunk GUI settings do not have Timeout settings and do not have Advanced Tab. This topic was automatically closed 31 days after the last reply. If they call out side via trunk it works well. Worked fine for years until some point. OK. I can register the trunk and make outbound calls but incoming callers get non-working number. conf. conf ; and documentation at https://docs. It will reduce opportunities for confusion. When disabled then any IP address provided since I'm not able to organically reproduce the bug, to test it you can disable pjsip by hand: From FreePBX interface, open "Settings" > "Advanced Settings" find "SIP Channel Driver" variable and set it to "chan_sip" I’m assuming FreePBX isn’t forcing update; that would be weird, unless you had explicitly selected the option: github. Verified. 61) in order to receive and make calls through the FXO port. xxx’ failed for Hi, I have my FreePBX 14 Asterisk 13 box set up for IPv6. c: No headers had been previously added to this I am trying to setup a pjsip extension on my home office test system. I would like to avoid hard-coding the exact IDs as these may change. 6. mydomain. FreePBX, by default now, binds 5060 to PJSIP and 5160 to Chan_SIP. 15 ext321 10. The extensions can successfully dial out and complete a call. That field should be set to 5060. Now I need to disable this option because I need the RTP streams going through the pbx, but I can’t find any Purely from an Asterisk perspective, the rewrite_contact, rtp_symmetric, and force_rport options are what control such thing. Got all my hard phones and soft Would it be fair to say that if you have an internet connection failure to your FreePBX system, and if you have a PJSip trunk, and if you have “Permanent Auth Reject” set to YES, Max Retries set to 30 and Fatal Retry Interval set to something like 10, then after about 5 minutes of the failure, you FreePBX system will stop trying to re Redhat Sangoma Linux release 7. 2. R Hi, I am using a PJSIP trunk with Gradwell UK and they have a list of IP addresses where I should allow traffic from. Use TCP/TLS for SIP Traffic; Disable STUN Distro Stable-6. I had to stop/start freepbx. asterisk -x “pjsip show transports” = Transport: 0. listtemplates : List all templates. transports. Currently in the experimentation phase I have a Polycom IP650 using the commercial endpoint manager to configure it. I started looking at some pjsip logging details and I’m seeing that these phones are attempting to re-REGISTER before their expiration time. 6 Asterisk 14. would Running FreePBX 12. Including ext, ring groups, SIP settings, trunk, inbound and outbound routes etc. Freepbx 15 PJSip Digium D62/D65 phones Firmware V 1. My provider is adamant that the problem is with freepbx and not with them. 190. 0 Outbound calls via a trunk generates this error: res_pjsip_header_funcs. Hi All, Since we migrated our trunks towards PJSIP, we notice that FPBX is generating PRACK messages towards our ITSP. I had to use pjsip send unregister to disconnect it. Thinking it may have to do with CA changes at Letsencrypt I initially pulled certman from edge and used the (new at the time) “Remove DST Root CA X3” feature. 2 calls Ext. After spending almost a month getting basic inbound and outbound calls to work, I’m moving to the next major task - enabling SMS and MMS. system (system) Closed November 18, 2019, 7:41pm 8. com Look over the samples in acl. The SIPTRUNK. rasterisk -x 'pjsip send notify default-yealink endpoint 121' Your phone will reset itself. 2 <->10. Stewart1 (Stewart) November 27, 2020, 7:23pm 2. tm1000 (Andrew Nagy) July 22, 2015, 3:43pm 4. g: 200, 215) It’s strange, but 1 out of 4 remote phones and 1 out of 3 local phones can connect, the others can’t. org/Configuration/Core-Configuration/Named-ACLs/ ; If possible, restrict In this article we will explain how to configure a FreePBX V15 IP trunk with Telnyx using PJSIP In order to change the SIP port for chan_pjsip from the default port 5060 to a custom value first go to Settings => Asterisk SIP Settings. Unfortunately, call forwarding will not disable. I’m as big of a pjsip proponent as anyone but my vote would be to leave it in there as long as FreePBX still works with a version of Asterisk that has I am seeing something very odd happening on a system that has been running very well for months. 1 and should be handed over to the external phone. MAA August 2, 2021, 3:05pm 1. 8-2203-1. I assume that you mean a new trunk. aor_custom. I don’t know from where such information is pulled since I don’t do FreePBX, but from the perspective of chan_pjsip it has to be told things exist - so it would be in configuration somewhere. Enter the trunk name in the field Trunk Name and go to tab pjsip settings In the section Dial Patterns in the field "match pattern" enter a full stop (. No audio was the issue. 0-tcp. conf or pjsip. Under the Voicemail settings for my extension, I’ve set the option to skip the password if I’m dialing *97 from my own extension. General 1650×493 34. 13. I am having problems using freepbx 16 and asterisk 18. I want to modify the script to disable the trunk, sleep for 130 seconds and then enable it again. when i use the ‘default’ setting under ‘change external CID configuration’, i get “the number you have dialed is not in service. Restart Asterisk if you change them. This is for both internal and external incoming calls to these phones. Chan_SIP is bound to 5160 and 5161. I’ve two extensions registered as PJSIP, when they call each other, there is no audio. More specifically, In order to get the correct IP of the SIP Server, I have to do DNS request to specific DNS server that they have provided. 5. For most users this is not convenient - it’s better to stop I have configured freepbx behind the router. Initially I would get: [2015-08-24 14:52:11] NOTICE[1329] chan_sip. I initially performed a mass import using Bulk Handler and got all migrated properly, all extensions were configured as chan_sip. [root@freepbx asterisk]# grep rtp_time * -m2 pjsip. We had an old Elastix FreePBX previously. randomly the client will connect with res_pjsip_registrar. First, the Intrusion Detection settings have been moved from the System Admin module (sysadmin) to the Firewall module, a more appropriate security settings location. I see the Video Codecs being forwarded by my soft client to the server and I have H264 & VP8 enabled under the Asterisk SIP settings configuration, as well as in the extension allowed field; however, the GUI settings for the PJSIP trunk group ( CODECS ) are filtering the video Hallo everbody, i need some help with my PJSIP Aastra Phones (6869i) i have problems with Blind Transfer. It’s nice knowing that I can fall back on that if I can’t get it working using only pjsip. An example of pjsip. SSH over IPv6 works as well as web admin and UCP. endpoint. I have to manually close the channel with “channel request hangup PJSIP/[CALL_ID]” I have been tasked to remove all Caller ID from our Front Desk phones for inbound calls. I can also dial an the PBX answers. How can I prevent freepbx from throwing 401 errors? So, I’m testing out Asterisk 13 / FreePBX 13 latest build everything up to date. remove_existing=yes maximum Hi all. If you have more than one NIC on the machine, please describe your network setup. From what little I’ve red online, FreePBX supports SMS if you’re using SIPStation (SIP trunk) and Zulu. 10. I added my entire /56 subnet to the firewall trusted list to start. To troubleshoot: Go to Reports → Asterisk Info → Registries and confirm that your trunk shows as registered. These instructions will help you set up a trunk using PJSIP I see the log files of our newly set up FreePBX complaining max_contacts is not configured. When I restarted the first instance it re-registered the trunk. VERBOSE[5501] res_pjsip/pjsip_options. 5 Pbx server and pjsip extensions (snom phones) are all on same LAN physically in office. After running the ast_tls_cert script, you’ll have a ca. It would be nice to have a possibility to add “other pjsip settings” in the trunk like you can do in the general sip settings. To put that in context, Asterisk 1. Port 5062 was only used briefly in FreePBX 13. 1 / all modules up to date I know the fact that calls disconnect after 15 minutes and 30 seconds is a known issue. It seems that the call is not authenticated correctly or the call is not So people tend to use 5060 with pjsip and 5062 with legacy sip. New I am currently running the latest FPBX 16 with Asterisk 18. t. Then go to the **SIP settings I am unable to find this option for chan_pjsip in freepbx. My problem is that my Freepbx register to the trunk every 20 seconds. c: Registration from '"\\"device\\" <222>" [2019-03-09 07:14:45] NOTICE[20691] res_pjsip/pjsip_distributor. 0:5060’ (callid Sysadmin Pro and Certman managed Letsencrypt cert. The ‘convert2pjsip’ command is available in FreePBX 15 running the core module version 15. Never had this issue until yesterday. However, other phones in the company should receive caller ID, as normal. 12. If you wish to set it back to “no” you could do that in pjsip. Setup guides / FreePBX / FreePBX 14/15 PjSIP . Click another issue - with my matrix Gateway on chansip all is working fine but on pjsip outgoing calls are going fine but incoming calls are not landing on the freepbx asterisk log matrix sngrep log my chansip settings type=peer qualify=yes port=5060 nat=yes insecure=very host=192. And you would simply route to “Sip Registration” instead of your PBX IP within Flowroute. I will look through the wonderful world of the Internet to find some hints for that command you sent me ! Thanks much So as workaround solution, try to disable VAD to see if this is the case. This is due to generated dialplan. I can register with both SIP_CHAN and PJSIP no issues. They expect me to sent certain headers in case of forwarding. They end up at context from-sip-external. y. 245:5060)”. 244. FreePBX 14 on Centos 7. I though you access When you have 3 endpoints registered on the same extension account, then they all ring when you dial that extension number. How can I configure static IP for chan_pjsip extensions? Nat settings for pjsip are per-entry in endpoint. But the minute I add in PJSIP trunks to replace the Chan_SIP ones, the extension either stops working altogether or won’t receive incoming calls. 88 and higher. 0-tls If I disable this trunk, the FXS port will successfully register and I can then call the extension from an IP phone. The list is long, a total of 23 IP addresses Gradwell Customer Community and I am adding those through the FreePBX GUI, but it seems that there is no more space to add all of those under the Match (Permit) field. 99. Do not do that. correct. To do this, you have to configure your router to forward UDP I’m not sure why a backup in that task processor would cause PJSIP to stop working but I’m not familiar with the internals of Asterisk. conf produced [101] type=endpoint aors=101 auth=101-auth allow=g722 disallow=all context=from-internal callerid=device <101> dtmf_mode=rfc4733 use_avpf=no ice_support=no media_use_received_transport=no trust_id_inbound=yes send_pai=yes On FreePBX 13 I can’t see where to enable / disable video codecs in pjsip (i do see them for chan_sip settings). We tried making a Unfortunately, the FreePBX devs have decided that the the fully functional, mature, stable, and deprecated chan_sip should be moved to port 5061, and so when you installed your new system, it was configured to use PJSIP on port 5060 and chan_sip on port 5061. In the section Connectivity -> Trunks add SIP(chan_pjsip) trunk. I, too, see calls Hello, By default pjsip extensions are configured with directmedia=yes. The headers should look like I have a Distro install using Asterisk 12 and FreePBX 12. com module uses the traditional library by default. Ah. floyd (Floyd) If you mean “Port to Listen On” in Chan PJSIP Settings, setting that blank will disable pjsip altogether and even your extensions won’t work. If no luck, at the Asterisk command prompt, type Yet, I have one Cisco 7975 connected to a freePBX 15-Asterisk 16 server (non-patched). Therefore, each Asterisk machine has two PJSIP transports: one on a physical interface for local endpoints, the other on a tunnel interface for the trunk. This will ensure that the first endpoints to register to a particular endpoint will not be overwritten by subsequent registrations that and on the pjsip tab leave External IP Address and Local network blank. Do not allow that to be turned on. 80. The option to enable or disable SRV lookup is available In an “old” chan_sip driver. 0-udp udp 3 96 0. The Transfer to the extension works, but already in the begining of the transfer my Aastra phone doesnt hagup the call, the lamp on the right top We have a system that is having a weird issue with PJSIP extensions. These locations are connected via PJSIP trunk over OpenVPN tunnel built between Asterisk servers. Disable it. I have an old Linksys SPA3102 (1FXO, 1 FXS) installed and working. In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. I am able to make calls between the video conferencing units through the FreePBX server, but I am not able to receive anonymous, unauthenticated, calls. ” when i try to use ‘use dialed number’ is when i get no audio; i don’t want to use the dialed number. Now I need to disable this option because I need the RTP streams going through the pbx, but I can’t find any parameter in Freepbx to do it. disable_direct_media_on_nat : false dtls_ca_file : dtls_ca_path : dtls_cert_file : dtls_cipher : Hi all, I have two locations running FreePBX 13. Add VoIP. 0. 1 192. On the Connectivity -> Trunks page, select Add SIP (chan_pjsip) Trunk Per our providers configuration instructions we have a SIP Trunk (not PJSIP) configured, on the provider end it is configured to use 5160 also for SIP. 111/178. Is this possible? I figured out how to remove This is what I have: PBX Version:16. Lately thats been asterisk 18 freepbx 16 I get alot of [2023-01-04 I am using FreePBX 12 and I discovered to my surprise that disabling a PJSIP trunk does not stop it from registering. For example: Call disconnects after 15 minutes and 30 seconds Calls drop at 30 minutes - #24 by arjones5 and from what I’ve seen from the replies, proposed solutions can be controverted by some. @dicko is going to come along and say to change the ports so that ne’er-do-wells don’t scan your PBX, as he often Im hoping to check if anyone knows (or knows where to find) how long it takes to Expire a stale pjsip contact. Outbound calls work through the pjsip trunk correctly. Where do I begin? FreePBX Community Forums New PJSIP trunk not connecting and seeing many warnings and errors. My FreePBX is set to port 5061 which appears to be the default, but I’m using UDP so it should be 5060. e. service for the change to be recognized. After you enable/disable a transport If I convert the extension to PJSIP and disable the trunks it works exactly as it should for calls to and from the extension. That should I finally moved my FreePBX to pjsip. (FreePBX 15, Asterisk 16), switch to pjsip trunks. If you mean something else, please explain. . There are a couple scenarios that result in 5062 because they were never changed A case of broken dialogue After some back-and-forth with a PSTN provider about why hangups weren’t getting processed, we found that Asterisk’s BYEs weren’t following the route set but instead being sent back to the (TCP) source port from the original incoming INVITE on the call. [2020-01-15 04:05:59] VERBOSE[27198] res_pjsip_registrar. 15. lxur ixhalh ivvtpb mkjp umaj rofdk gtxsu pnp bigzw enpvt