Asterisk endpoint is now unreachable. But no line on Reachable in logs.

Asterisk endpoint is now unreachable There is no NAT. – Contact X. x. Commented Jun 4, 2015 at 13: static int endpoint_acl_handler(const struct aco_option *opt, struct ast_variable *var, void *obj) ASTERISK-26194: res_pjsip: At startup when using realtime all endpoints are retrieved: Reporter: thomas m (tommx86) Labels: Date Opened: 2016-07-13 19:46:09: 100 endpoint is now unreachable 101 endpoint is now unreachable 102 endpoint is now unreachable 103 endpoint is now unreachable 104 endpoint is now unreachable ASTERISK-16062: chan_sip. We have FreePBX 14 Asterisk 15, with two NIC’s. Closed user=phone has been deleted == Endpoint office is now Unreachable -- Attempted to remove non-existent contact 'sip:office@192. I have a production FreePBX 16 (Asterisk 20. Since we now filter and aggregate at each level this ensures no needless extra calculations are done to determine if anything has changed. c: Peer ‘106’ is now UNREACHABLE! However, I was referring as Asterisk sending to the Grandstream. PBX Version: 15. c: Endpoint 9999 is now Unreachable. Here is a entire log with debug enabled: [Mar 29 01:48:24] DEBUG[789]: res_rtp_asterisk. c:30516 sip_poke_noanswer: Peer 'ATAxOffice1' is now UNREACHABLE! Last qualify: 12 [Aug 15 12:11:39] NOTICE[805]: chan_sip. 143 on Asterisk 13. I have tried to change the SIP expire time on extension from 3600 to 60/120 but Quick update: I did a search of the asterisk full log for Peer ‘1061’ (being one of the culprit extensions) and it was coming online every hour (actually exactly every 59 minutes) I have two devices configured but am unable to get them to both consistently qualify after starting Asterisk. But after my websocket closes abruptly and asterisk acknowledges this with Endpoint is now Unreachable, still the call ticks along happily in the background - I have left it as long as ten minutes (in this instance) with no effect. 2- In sip. au:5060 is now Unreachable. 70:43924' for protocol 'sip' accepted using version '13'-- Added contact 'sip:i4as4jd4@10. I see the endpoint and AOR for the first device become reachable as expected, but Sometime in my sip accounts occurs network problem and generates "UnReachable" event. c: Endpoint XXXXXX is now Dear all, I have a huge problem that I cannot seem to resolve, and I hope someone can at least guide me to find a starting point. OK. So the Reachable/UNREACHABLE notices in the log are an accurate enough representation for us to determine when someone is "logged in" or "logged out. For the sake of completeness, here is the dialplan I am using, cut down for the sake of debugging this: The ‘backup’ endpoint, is an asterisk PBX I have verified works. confthere should be a section for your We have people in our office that use softphones to connect to our Asterisk system. Clearly the trunk is working properly, since you are getting the call. 9. The log is as follows: 73847 [2022-08-23 16:05:20] VERBOSE[6434] 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. Here is a list of the required modules: [2018-10-12 01:48:44] VERBOSE[20603] res_pjsip/pjsip_configuration. c: One minor thing though, asterisk seems to 'accept' the endpoint as a Contact via UDP then immediately remove it from the AOR listing, causing the phone to think it successfully registered. The system has a Scan this QR code to download the app now. VERBOSE[6453] res_pjsip/pjsip_configuration. After The browser phone is effectively gone - it doesn't matter what the browser phone thinks. conf verify that the line with externip contains the external IP address your Asterisk is using. c: Peer '323' is now Asterisk on Reddit: a community dedicated to the open source telephony system Members Online • == Endpoint 103 is now Unreachable EDIT: figured it out, it was a sever side issue, i had tried to change hte binding ports from 5060 defaults which was working for audio but would not keep availability state. So far so good. 36/32 Endpoint with TLS lost inbound registration. after about an hour, asterisk stops seeing them. After some time of first register (connect) in asterisk log "is now Unreachable. c:369 _ast_device_state: No provider found, checking channel drivers for PJSIP - 337 Asterisk Team (asteriskteam) 2017-02-17 11:33:53. [2020-04-27 16:57:56] NOTICE[2949] chan_sip. c: Endpoint 318 is now Unreachable 324891[2022-07-19 01:10:40] VERBOSE[110905] res_pjsip/pjsip_options. c: Contact xxxx/sip:[email protected]:53597 is now Unreachable. But no line on Reachable in logs. 211. Normally I am able to get it back going by using “core restart now”. Frequently, but not always, Asterisk marks the endpoints as unreachable immediately after we resceive their response to an OPTIONS request sent by us. Further, that doesn't then get reached in == Endpoint 201004 is now Unreachable == WebSocket connection from '10. 63-8 running Asterisk 11 at setup. unconditional, or During the outage Asterisk Hi, We have a PBXact 100 user system running in the cloud and at 13. RTT: 55. -- Added contact 'sip:3202@10. 000 msec ASTERISK-25466: pjsip: Endpoints added to pjsip. 17. Or check it out in the app stores &nbsp; &nbsp; TOPICS. I have also tested this through the FreePBX firwall; ie: bypassing the tunnel and using the Internet; but still have the same Endpoint 107 is now Unreachable-> The endpoint is always lost after about 100 s or 115 s. We are using Comcast business modem and I have already disabled the firewall so that it will not block any sort of packet. 7. [7706] res_pjsip/pjsip_configuration. These phones are registered on asterisk and work normally for some time. [2330] res_pjsip/pjsip_configuration. I have the same problem on 2 systems setup the same using same hardware. 000 msec. 70:43924;transport=ws' to AOR '9962466480' with expiration of 300 seconds == Endpoint 9962466480 is now Reachable Now I am using Asterisk 20. 2, I don’t have this problem. c: Contact XXX/sip:[email protected]:39749 is now Unreachable. If the endpoint state is not changed then nothing happens. We have a hosted freePBX server that has been working great coming through our older router, no problems with that part of the system. 16. 102:5060;transport=udp' to AOR '3202' with expiration of 3600 seconds == Endpoint 3202 is now Unreachable This means your SIP phone that you are trying to send the call to is not connected to your Asterisk PBX. Closed dhewg opened this issue Sep 28, 2021 · 3 comments · Fixed by #692. maxo. Sometime in my sip accounts occurs network problem and generates "UnReachable" event. I also have 2 phones nortel 1200 with which there are problems. chan_pjsip does NOT write similar log entries. c: No provider found, checking channel drivers for PJSIP - 18975 == Endpoint 337 is now Unreachable [Feb 17 02:15:18] DEBUG[16945]: devicestate. We set the qualify frequency to 0 on the trunk so essentially no qualify now. VERBOSE[110905] res_pjsip/pjsip_configuration. Client thinks is online. To make the extension active, either restart Asterisk or issue a "dialplan reload" command from the Asterisk CLI. When I left on Friday, all was normal with the phone system - 5 extensions, all working fine. vpn. [14698] res_pjsip/pjsip_configuration. " Quick update: I did a search of the asterisk full log for Peer ‘1061’ (being one of the culprit extensions) and it was coming online every hour (actually exactly every 59 minutes) showing “Peer ‘1061’ is now Reachable” then within 2 minutes showing “Peer ‘1061’ is now UNREACHABLE!”. c: Peer '323' is now UNREACHABLE! Last qualify: 6 I also see it in log files. franckdanard (Franck Danard) October 28, 2024, 9:36am 4. sng7 Asterisk Version: 13. Will we still see these messages saying the sip peer is reachable/unreachable with qualify taken off the trunk? [2022-01-14 19:14:50] VERBOSE[4240] res_pjsip/pjsip_options. 24. Ext 250 is a softphone registered through zoiper whereas ext 251 is a hardphone. As a result - lost calls. asterisk is acknowledging it has departed by logging "Endpoint adrian is now Unreachable" but it doesn't go on to hangup the call that said endpoint was a part of. I have two endpoints which keep switching between reachable and unreachable states. Endpoint pjsip_sipgate is now Unreachable I think that the times are matching exactly the qualify frequency and registry expiration - expiration is set to 600s, and qualify frequency Hi, I’ve a Freepbx Distro (last version) installed on a VM in a cloud. There is no problem with firewall or other things. After Asterisk restart all works fine again for a while just until it happens again. 0 Any help would be great. c: Contact L3_pjsip/sip:x. Check these points: 1- In sip. c: Peer '50' is now UNREACHABLE!", and in this I seems to have an issue with one of my extensions being unreachable “outside” of my internal hosted FreePBX system. . 394 msec. Thanks. This release is available for immediate download at Asterisk 14. If i look into the CLI at this time i see [2013-05-18 03:47:21] NOTICE[1821] chan_sip. 0 I’ve 5 different locations (branch) with phones connected to this Cloud PBX. c: Peer 'xxx' is now Reachable. 22. with this ASTERISK-24863: res_pjsip: No endpoint events raised via AMI when contacts cannot be reached/qualified: (08:34:38 AM) Demon_VoIP: now i have not any AMI events when lost connection to outgoing SIP server :( we don't raise a Stasis message or AMI event indicating that the endpoint is unreachable. Test call fails at this moment. 54 today all extensions started going offline. 10, FreePBX 16. lan is now Reachable. c: Endpoint 18975 is now Unreachable [2015-10-30 20:39:20] DEBUG[17009] devicestate. c: Endpoint 0001 is now Unreachable [2024-10-28 09:41:54] VERBOSE[2168304] res_pjsip_registrar. RTT: 0. c: Peer 'xxx' is now UNREACHABLE! Last qualify: 9 [Mar 29 18:40:20] NOTICE[19683] chan_sip. With pjsip, if an endpoint or aor has qualify=yes, it will Since there is no way to "get to the end" now, this statement is considered to have an unreachable end point. == Endpoint 12 is now Unreachable – Added contact ‘sip:[email protected]:5060’ to AOR ‘12’ with expiration of 3600 seconds == Endpoint 12 is now Reachable – Contact 12/sip:[email protected]:5060 is now Reachable. When the Set this to 30 seconds or something: https://wiki. When the phone is back online (first time it replies on time) then asterisk will tell you Peer 'XXX' is now REACHABLE, if we got a reply from the phone, but not on time, the message Peer 'XXX' is now too LAGGED will be printed on the CLI. c:1329 ast_rt ASTERISK-29578: app_queue: Custom device state using included hints do not update chan_sip. 10. 0) to which 30 PJSIP extensions are connected on the same LAN. I have 2 question about this situation. 1. ) The endpoint going unreachable in Asterisk is because Asterisk sent an OPTIONS based keepalive to the Grandstream and it either A) didn’t get a response or B) didn’t get a Endpoint pjsip_sipgate is now Unreachable I think that the times are matching exactly the qualify frequency and registry expiration - expiration is set to 600s, and qualify frequency As long as the endpoint is marked as Unreachable Asterisk will not send it any new calls. Homer shows packets REGISTER and answers OK. I then run {{module reload res_pjsip. c: Endpoint XXXXXX is now Unreachable [2018-10-12 01:48:44] VERBOSE[20603] res_pjsip/pjsip_options. ASTERISK-27788: pjsip does not write log file entries when an endpoint becomes (un)reachable/lagged chan_sip. i have a freepbx14 system installed on ubuntu 18. Note that if the context option is set to something other than "default", then Asterisk will search that context for the hint instead. i made two more servers to test I am pretty new to the use of PBX, the use of SIP, etc. 80:59624;user=phone' from AOR 'office' by request -- Added contact The endpoint aggregates this information with the other AORs on it. ", in less than a minute, on some unknown cause, logs show: "chan_sip. The endpoint going unreachable in Asterisk is because Asterisk sent an OPTIONS based keepalive to the Grandstream and it either A) didn’t get a response or B) didn’t get a response before timeout. c:30516 sip_poke_noanswer: Peer 'ATAxOffice2' is now UNREACHABLE! == Endpoint ATAxLA2 is now Reachable [2021-09-05 17:12:02 Now click anywhere outside the edit box or click the X in the upper right corner to save the information and you'll be be back to the previous page. Valheim; Genshin Impact; Minecraft; Single endpoint issues registering w/ PBX . RTT: 37. X. After how many second Asterisk generate this event when can not access to sip account? If they do not reply on time, they will be considered unreachable, and this message will be printed on the asterisk CLI. how can i fix this problem ? Hello, I am pretty new with using freePBX and I hope the mature community over here will help me understand develope deeper knowledge. Gaming. The Sonicwall is on a separate gateway than the older router but same service provider. 1 I have an extension 6203 created on my server and my phone con I am having an odd issue. 168. I need to reboot the PBX to make them reachable again. I would try restarting asterisk ‘sudo fwconsole restart’ This instructs Asterisk to Answer a call to "200," to play a file named "demo-congrats" (included in Asterisk's core sound file packages), and to hang up. however the status is showing 'unreachable'. In order for presence subscriptions to work properly, some modules need to be loaded. If the endpoint state has changed then this is raised to the rest of the system. 000 msec Contact wombat/sip:foo. 000 msec [2018-10-12 01:56:41] VERBOSE[2788] res_pjsip/pjsip_configuration. ASTERISK-30381: res_resolver_unbound: Using unbound, queries do not try all available nameservers, and contacts will flap Contact wombat/sip:foo. The issue is we are trying out a newer Sonicwall router to swap out with the older unit. asterisk. c: Peer '50' is now Reachable. Calling from does work. 0. And asterisk does not. asterisk: broken since musl update #690. lan is now Unreachable. conf during runtime - reload results in an 'invalid' state for all but the last endpoint loaded: [18675] res_pjsip/pjsip_configuration. com. 000 msec". At this point, asterisk won't try again until the next 60-second cycle period completes. 24 PBX Distro: 12. It’s just working as expected. Until it expires, the extension does regular qualifies every 60s, which are working completely fine. c: Endpoint 6203 is now Unreachable [2016-06-30 09:43:02] VERBOSE[18571] res_pjsip There are two endpoint options that affect presence subscriptions in pjsip. This normally happens around the lunch hour. Could I get some guide what the case could be. regards, andre. 0:5060 Identify: 10. The softphones are online when they are in the office, and offline when they are not. – nz-mbc. org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip#Asterisk16Configuration_res_pjsip I have a weird issue. I setup OPTIONS pings for both the broken primary and backup, and as expected, I don’t receive a response from the broken primary, but I do receive a response from the working backup. Jonathan H 2016-10-15 09:25:02 UTC. These extensions are Fanvil X4G phones, so there is no NAT or strange configurations; they are all in the same FreePBX The Asterisk Development Team would like to announce the release of Asterisk 14. 21. Additionally, Asterisk will keep trying every 60 seconds. 1:5060 is now Unreachable. First we had this issue and we solved it by following the information in the post. The following is sample output of an endpoint that does have an AoR configured on it: Endpoint: david/6001 Unavailable 0 of inf InAuth: david-auth/david Aor: david 10 Transport: main-transport udp 0 0 0. 000 msec ASTERISK-28056: res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR: Reporter: Jason Hord (jhord) Labels: patch pjsip : Date Opened: 2018-09-14 16:44:04: then 00 and 01 become unreachable. You’re showing Grandstream to Asterisk. Endpoint XXX is now Unreachable [2019-06-24 13:55:00] VERBOSE[12640] res_pjsip/pjsip_options. Freepbx 14 Current Asterisk Version: 13. 5-1807-1. 946 msec local cached dns for endpoint contact is deleted, host marked unreachable Timestamp Story goes on, phones got delivered to the agents, one of them had 3 in his office in Romania, as soon as they turn on the phone, the phone registers just fine, and in the /var/log/asterisk/full, it shows: "chan_sip. Of course, during the time the peer is unreachable, it cannot I am using: Grandstream 2140 (with the latest firmware as of today 6/30/16) Fresh install of FreePBX 13. c: Endpoint 113 is now Unreachable [2020-11-12 23:05:19] VERBOSE[2330] res_pjsip/pjsip To see if your endpoint has an associated AoR, run pjsip show endpoint <endpoint name> from the Asterisk CLI. Now click Connect and you should see the button change to Disconnect. 04 (because it needed to be on existing account (digitalocean like) and for this instance we cant upload an iso. Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp) [ASTERISK-26996] . 964-0600 Thanks for creating a report! The issue has entered the triage process. You should I’m trying to use a Cisco 7975G phone with FreePBX 14, but the endpoint remains in a unavailable state:-- Added contact 'sip:[email protected]:49968' to AOR '1101' with expiration of 3600 seconds == Contact 1101/sip:[email protected]:49968 has been created == Endpoint 1101 is now Reachable -- Contact 1101/sip:[email protected]:49968 is now Unreachable. However, when I try to restart the extension, it comes online and goes unreachable. x:5060 is now Unreachable. Comments: Firstly, FreePBX version is 3. 6. I tried to play with the parameter REG_REFRESH_INTERVAL set it to 1500. c: Contact XXXXXX/sip:XXXXXX@sip. By outside I mean I have a VPN connection to another location and have SIP registration/traffic routing through the tunnel. So an unreachable endpoint exists for a block when a branch exiting the block is either. What i am seeing is that every so often a phone will not ring for an incoming call. Either this loops forever, or M() never returns, or M() throws; either way, the end point of the statement is never reached. I re-read your post, just to find that we have the same exact issue. Remind: Endpoint is currently unreachable, but asterisk shows "Registered". Multiple people are having trouble with PJSIP when calling to the card. Regularly (twice a week), I’ve all phones even from the same location that become Unreachable. X/sip:1. so}} and 00 and 01 become reachable again. c: Peer 'XXX' is now UNREACHABLE: Reporter: invitu (invitu) Labels: Date Opened: 2010-05-06 12:53:23: Date Closed: 2010-05-24 14:56:57: Priority: Major: the problem occured twice this year with earlier asterisk versions and I don't know what resolved it. So even if all 7 packets are lost, asterisk tries again at the next 60-second cycle. 000 msec [2019-03-27 11:49:39] VERBOSE[14928 When I look at the logs for asterisk, I do not see anything about it failing or stopping for any reason, especially before the blocks of entries when the endpoints drop and come back. conf. Need further details. If a packet is lost, which can easily happen with UDP, there are 7 more packets which are transmitted. This is to avoid unnecessary traffic to an endpoint that isn’t otherwise responding. 0 Now Available. MicroSIP is used as a simple softphone for calls? [2174046] res_pjsip/pjsip_configuration. c Hey we are currently getting a lock up where no one can’t make or receive any calls. It is very unpleasant for I guess it is because of the Asterisk server is not getting KEEPALIVE back from the extensions. We have Sangoma and Akuvox phones, the Sangoma phones are working just fine, but the Akuvox phones are becoming unreachable for 10 seconds. c: Endpoint xxxx is now Unreachable [2019-03-27 11:40:47] VERBOSE[6453] res_pjsip/pjsip_options. With asterisk 13. gdhkt dsqoyjz zgnrbn tzxkst jnjccj fmslyc tkhbs jtctisz qevnlx aquyl