Freepbx rtp timeout. The same phone works fine on the LAN.

Freepbx rtp timeout 40. If someone has a more durable solution, I’m open for suggestions. 11. I wish to install an external SIP phone (Grandstream BT200) on a public internet address behind a NAT. ANd in the asterisk log i see no activity RTP since 30seconds (timeout). Currently we have a separate dedicated VLAN for VOIP also PBX server has two Nics one to publi IP configured and another was connected to LAN VOIP FreePBX 15. edit: as a workaround i could easily try to increase the RTP timeout to 40 secs for example ? PBX and both phones are on the same network. All of mine phones are either on the LAN, or are not NAT with a site to site VPN. 1 / all modules up to date I know the fact that calls disconnect after 15 minutes and 30 seconds is a known issue. la chiamata viene eseguita ma nessun audio, dopo 30 secondi RTP timeout. I create a backup via “Backup & Restore 16. c:29981 check_rtp_timeout: Disconnecting call 'SIP/100-00000001' for lack of RTP I am attempting to set up a simple Emergency Broadcast in our FreePBX system (FreePBX 13. I was able to set RTP timeouts on the endpoint so that it recognizes loss of connectivity and hangs up, but the call on the Asterisk server side of things continues indefinitely until This option work correct when call is not holded. However the call drops afterwards with the message Disconnecting channel 'PJSIP/23-0000005a' for lack of audio RTP activity in 33 seconds. Cricchetto (Dido Cricchetto) November 19 , 2016 Then in sip setting - chan sip I changed the parameters rtp timers, but still the call drop after 32 seconds. rtp_timeout_hold : 0 sdp_owner : - sdp_session : Asterisk send_diversion : true send_pai : false If you want to evaluate FreePBX/Asterisk as a whole for your business then using a forked/non-Digium/Sangoma supported version of Asterisk and trying to setup custom extensions for GVSIP (which is not support at this point by FreePBX) is a poor Over-aggressive Silence or “No RTP” Detection. Most firewalls handle RTP just fine (or as you say “sort of work”), without enabling SIP ALG, for the reasons you describe. 8x The RTP timeout value on the SIP settings page is 30 seconds. Setup Example: I am ussing some Unify SIP phone, and some softphones (android, mizudroid and PortSIP Softphone ). But every now and then there is no sound on all endpoints. It doesn’t however work when ring time is set to 20 seconds. Very few router/firewalls have proper SIP ALGs that monitor the SDP and open RTP ports correctly, while also avoid mangling the signaling. We originate from a local context to a pjsip and everything works fine 99% of the time, but every so often it looks like the channel gets hung up on somewhere yet the dialplan still runs until it gets to the AMD command, then stays there until the trunk provider terminates. I have many chan_sip clients cooperating flawlessly with Inbound and Outbound routes, ring groups, behind NAT etc including another IAX2 server and some SIP trunks in both of sides. 15. pjsip set logger on) demonstrating that Asterisk closes the call in response to the un-REGISTER, and not in response to a BYE, or session timer or RTP timeout. The FreePBX is a VM sitting on an ESXi hypervisor, thus it is using a virtual interface that is in a DMZ VLAN. 000 verso IP Asterisk. 24) and I am having trouble setting up a bandwidth. It might be a NAT configuration on my freepbx because if i try from different networks im getting the same issue. When the line set as primary drops, the FreePBX switches to navigate correctly on the second line and still registers correctly with the VoIP 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. is there a command I could type to force flush the queue? and RTP Settings: RTP Port Ranges: Start: 10000 End: 20000 RTP Checksums: Yes Strict RTP Yes RTP Timeout: 30 RTP Hold Timeout 300 RTP Keep Alive 0 Everything else until Codecs Blank Codecs ulaw, alaw, gsm, g726, g722, g729 Video Support: Disabled-----END Asterisk General SIP Settings-----START SIP Settings [chan_pjsip]----- Hi, guys, I have a freepbx setup with a Goip “sim gateeway”. 37. Many of the posts offer great suggestions, however, none seem to get my system back up and we have bandwidth. I have an installation of FreePBX 14. 240 context=from-internal I have FreePBX 2. The dial command is being given a 300 second timeout by FreePBX. com as our provider but not through freepbx because I ordered the trunks first and didnt know about the module. I disabled the keep alive, and now the audio is crystal clear. context: (not set) Regexten on Qualify: No Trust RPID: No Send RPID: No Legacy userfield parse: No Send Diversion: Yes Caller ID: Unknown From: Domain: Record SIP history I have been working with my provider on an issue we have with some calls dropping due to lack of RTP activity - here is the response from our provider I reviewed the call in question and found that in your SIP Allow headers it is missing the “UPDATE” in the list. I also wanted to implement secure trunking, which has worked, no problem. 4 phones: Cisco spa 303 Trunks: bandwidth. 1 The pbx runs on a server directly connected to the internet, behind firewall, accessed only from LAN. The extension at home using Grandstream Wave phone app is working well too. And Asterisk dos not terminate call after 11 seconds if no RTP or RTCP activity on the audio channel. 1 Like. On Asterisk or FreePBX systems try OK - interestingly enough. When an external caller hangs up during the announcement or during the following waiting period, the Ring Group still gets called. The issue is RTP is the audio stream. I’m 99% sure this is not a FreePBX issue, it’s going to wind up somewhere in the firewall I’m sure. Randomly there are extensions that without any pattern stop connecting to the RTP ports. conf but that doesn’t help either) vieri March 26 I am having a problem where millions of tasks are queued up and then the PBX stops responding. This server has a public static IP; therefore I can access both FreePBX and Asterisk on that IP with no problem. We have always set up using Public Server Wizard and left those NAT rules as default, allowing ports 5060-5061, as well as the RTP ranges requested from ITSP and locking it down to only allow Hey guys im new to freepbx and im trying to build an extension that can make and receive calls, which should be pretty straight forward(or so i thought). I had to do an fwconsole restart to get back to normal. I was able to set RTP The apparent fix here is to add in an RTP keep alive, which is in "Settings > Asterisk SIP settings > Chan SIP" and in the "rtpkeepalive" value I've added in 10 initially to test. We had users on a new phonesystem (Part of an existing one, but using a new call server, running latest Freepbx 6. dotcom I am constantly seeing this in my asterisk CLI. c:29611 check_rtp_timeout: Disconnecting call 'SIP/400-000030f4' for lack of RTP activity in 31 Hello all! We use FreePBX to, amongst other things, connect users to audio couplers over POTS. As soon as I set my Time Conditions to ring the IVR’s I have setup and call my number all I get is silence. On the Grandstream “Advanced Settings” page, the default RTP port specified is 5004. RTP Hold Timeout is 300 and RTP Keep Alive is set to 0. It binds to them dynamically. Is there some overall timeout that is causing this at 20 seconds? going to try each ring group at 5 I ran a packet capture on our firewall and can see the initial ring in from my remote extension (cell phone sip client - not behind another NAT) The PBX acknowledges it and sends it back on port 5060 and the call is established. Network Firewall Configuration I have reached the end of my tether, could Re posting this as another thread was closed. I am using FreePBX 2. RTP settings FreePBX: RTP start 10000 RTP end 11000 timeout: 30 RTP Hold Timeout 300 RTP Keep Alive 0. org and post the link here. B / the local network is 192. 35 / Asterisk 11. 0) SDP Session Name: Asterisk PBX 11. : false rtp_keepalive : 0 rtp_symmetric : true For RTP traffic Asterisk doesn’t bind to 10000-2000 ports on startup. We are currently advising that the customer modify their UDP session timeout on their router for port 5060 to be 45 minutes while the phones register every 30 minutes. (I mentioned RTP flow because I also set rtp*timeout values in sip_general_custom. I can call the softphones from IP ToS RTP audio: EF IP ToS RTP video: AF41 IP ToS RTP text: CS0 802. 7. This is in regards to FreePBX 13. A. endpoint. AsteriskNow is not the best FreePBX based system so you might as well move off of it now. With this route 10. I am getting rtp timeout errors. The same phone works fine on the LAN. Hello I have successfully installed my freepbx 14 on vultr account. 49 (Asterisk 13. Configuration. 3/24 in the advanced sip setting. Instead the parked call is dropped. 0 gw 10. What can I do? I used freepbx version 13 and asterisk 13. Sound about right? johnjces (johnjces Good afternoon, I’m just learning asterisk. New replies are no I deployed FreePBX in a docker compose. The caller ie cell phone hear’s no audio till he call is put on hold and then they hear everything hello And I have a problem with a single extension, the 300, it makes a normal call, records the audio, but does not receive a call. 2. 1. computermikes (Computer Mikes) July 20, 2021, 5:53pm 3. 729 licenses installed. claloano (Claudio Pelosi) May 25, 2020, 9:34am Disconnecting call ‘SIP / 0108933605_in-00000074’ for lack of RTP activity in 31 seconds. dpedersen13 (dpedersen13) February 25, 2024, 2:40am 21. Calls are get delievered to a private IP space from level 3. I can’t get ports to register any longer now that all of the communications has been moved to pjsip. I set the RTP Timeout to 180. conf:rtp_timeout_hold=300 IMO most of the systems out there need it. c: Retransmission timeout reached on transmission with my FreePBX configuration. Networking. 65” I install a second Freepbx 16. Even without registered endpoints or calls it takes hours for the queue to empty itself. 33 System Version 12. That means sip is on port 5060 udp and rtp is on 10000 to 20000. xxx. It works fine in local but when I go through internet, I could make call but there is no sound ! I open port to forward tcp/udp 5061 and tcp/udp 10000 to 20000. Issue is with DISA. We are originating a call from one context that does a dial to another. Hi all, We experience an issue where a Queue agent answers an inbound call but does not hear the caller (silence). conf config file to set the qualify_timeout to 6. you have admin access to the PBX, look for settings that reduce the sensitivity during DTMF detection. If I setup up this way: Incoming call -> Ring group B -> IVR, it works when the ring timeout is set to 10 seconds. looks like you are experiencing " for [2024-12-05 16:09:58] NOTICE[8005]: chan_sip. If there was some way to limit the length of the Good afternoon, We have installed FreePBX 2. 0 SDP Owner Name: root Reg. 210) & 24 (IP: 192. Hoping to get some help from esteemed experts; I am new to IP telephony and am trying to set-up a FreePBX distro. Packet timed out after 11200ms with no response [2014-02-13 14:32:37] WARNING[1712]: chan_sip. This process works great when a call from a softphone (e. Asterisk terminate call after 11 seconds if no RTP or RTCP activity on the audio channel. Just setup fanvil x3sp phone as local extension FreePBX is local on same lan etc Shows as registered on phone Can dial out However FreePBX shows as unavailable and unregistered Any ideas Works as extension with another voip service (external) rtp_timeout=30 rtp_timeout_hold=300 rtp_keepalive=0 send_pai=yes rtp_symmetric=yes rewrite_contact Any help to point me in the right direction would be greatly appreciated. RTP Keep Alve: 1. Here is a simplified I have a FreePBX version 14, the calls from outside to inside fall in 32 seconds, regardless if there is voice or not, I researched it, checked all possibilities plus nothing at all, my FreePBX NAT identifies the external IP normally, the RTP ports 10000-20000, SIP 5060 are open, I put in sip. My phone screen still shows that 9 when the call is connected. Last Thursday, I received notice that suddenly, incoming calls are getting dropped after ~30 seconds. So far, setup of a Digium card with FXO port and installation of the distro itself was cake. 26:0 In the SIP settings section, in the PJSIP settings, the keepalive interval is set to 90. lgaetz (Lorne Gaetz) May 13, 2021, 1:16pm 3. 4, freepbx 2. 49. NOTICE[1734]: chan_sip. here RTP is the audio stream. Also we are able to hear both side voice. X D A I have recently built a new FreePBX server running FreePBX 16. I’m using RasPBX running FreePBX 15. In Freepbx, see if there's RTP Keepalive or other timeout values. c: Retransmission timeout reached on transmission 739107163_73741112@4. I looked at the asterisk console in verbose mode as I was calling in I have been pulling my hair out over this and I’m hoping you guys can help me out here. Ho messo la regola RTP 10. Also as previously stated the issue happens across multiple carriers so appears to be switch related. So i installed freepbx on an aws server and have no errors, i created an extension which does connect to zoiper and is active but when i try to make a call to my phone number i get a lady saying “services are busy” and One of our customers has a (relatively) new issue when forwarding calls. Any input would be greatly appreciated as I’m totally stumped at this point. t How I solved chan_sip. I hope someone will help me. 9. When configured, channel 0 will use this port_value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. 9) I have configured my firewall to allow asterisk’s ports below: 5060 (tcp) 5060 (udp) 4569 (udp) 10001:20000 (udp) 5038 (tcp) my problem is: when call comes in, I am unable to hear caller’s voice!, but the caller able to hear my voice and then the call was dropped after 30 seconds In Asterisk SIP Settings, try setting RTP Timeout to 180 (3 minutes). However, when I attempt to transfer the call on the MicroSip, the call is not transferred. FreePBX Community Forums Disconnecting call for lack of RTP activity in 31 seconds. My sip is behind a nat with a static ip adress i have tryed all configs i can chance on my frepbx server. I have done the necessary initial setup including the firewall setup. If you telnet to an asterisk system on UDP port 10000 and there are no calls happening you will just timeout. c: Disconnecting channel ‘PJSIP/7450-00000016’ for lack of audio RTP activity in 30 seconds For some time I can hear silence from 7450 without stopping time counting. c:29115 check_rtp_timeout: Disconnecting call ‘SIP/vitel-inbound-0000008a’ for lack of RTP activity in 31 seconds == Spawn extension (ivr-2, s, 10) exited non-zero on ‘SIP/vitel-inbound-0000008a’ I have another FreePBX system that my office uses (5 extensions) and I’ve compared the IVR Request Time out (408) phone on internet ,Freepbx behind OPNsense firewall - FreePBX. These are remote extensions and don’t go through the regular trunk. NOTICE[12142]: chan_sip. 10. Call Drops but workable on Linphone(Desktop) to Linphone(Mobile) I have been checking: 1. The trunk is registered and inbound routes and outbound routes are configured and working properly. 0. 1) in the Paging and Intercom module (13. I need to lookup how to change UDP timeouts for that router. The phone works fine and I can make calls. 254 internal My default gateway is currently set at 10. 4. They are in rtp. org) specifies RTP 10000-20000. C183VM4U*CLI&gt; sip show settings Global Settings: UDP B&hellip; Hello All, I have problam to make outside calles from Skype client but i recieve call. That is way above the 30 seconds FreePBX uses for the default RTP Timeout. If the call passes through IVR or a Hi everyone, I have installed FreePBX 15 and Asterisk 16 on one of my virtual servers running CentOS 7 64-bit at Linode. In chan_pjsip it is “rtp_timeout” disabled by default (in Asterisk) and in chan_sip it is “rtptimeout” also disabled by default (in Asterisk). 13 on CentOS Linux release 7. 8 running on a beagle bone black (www . I’ve configured the trunk and an outbound route and I can make outbound calls OK. Please help me. This will keep the session open so an inbound call won’t be blocked. Hi, We have 1 FreePBX server in 1 office and we have 3 offices in total. FreePBX Community Forums Is there a max amount of seconds for Park timeout? FreePBX. My questions are: Can Hello all with the Help of avayax on the Grandstream FXO gateway trying to connect to it. I bought paging pro for it’s valet feature. 16. Pastebin link: https:// pastebin asterisk and the RTP timeout This was an interesting one after a new install of Asterisk (using freepbx) and one I wanted to document and log for future. It would seem an RTP communication problem. While I migrate to different hardware, I am having to restart asterisk and/or the complete machine several times per day. nielsen April 12, 2021, Hello, I have 4 asterisk servers and I need to create a SIP trunk between them. 40 with just one extension. 200. They have stated: Sparklight reserves the right to employ network management practices (e. Hi, getting complaints about dropped calls. I am having problems getting calls into my freepbx. freepbx. Endpoints. dotcom (dotcom) August 8, 2022, 2:41pm 1. I had found connection timeout errors in the One-way RTP Stream mid-call??? Polycom 321 Hello, I have a strange problem where the RTP stream will stop mid-call. below, I when i connect, on the cli i have : [2014-07-31 15:01:58] NOTICE[6839]: chan_sip. 0) and finding that putting people on hold, the Here is my environment: freepbx and asterisk running in a docker, at a simple ubuntu machine on azure. When i receive or make a call from outside on one of our extensions, I cannot hear anything but the other person can hear me. Can anyone give me some insight on what’s going on based on this log clipping? Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: FPBX-13. Container works fine and SIP extensions links with the devices sucesfully. NAT Configuration 2. When a place a call comes in from outside, if the call is routed directly to the sip device there is no problem. Set the UDP Timeout on your LAN->WAN Firewall Rule to 300 seconds - the default is I am looking for your help: We are using FreePBX 14. I have updated the phone to the latest firmware (4. PeterFox (Peter Fox) February 10, 2021, 9:47pm 1. But I I’ve just installed FreePBX Distro on a client’s system and everything has gone swimmingly well with the exception of the timeout of parked calls. They aren’t registering with a Request Timeout (408). Hi, i have a trunk on my freepbx behind NAT that continuosly disconnect calls on 30 secs (just rtp timeout) what to do? FreePBX Community Forums RTP Issues with ChanSIP. When i call between extensions, call is made but finish after 30 seconds because in RTP setings i declare 30 seconds in RTP Timeout. It does still drop the call when it hits the “RTP Timeout: 30” so it’s obviously still not properly seeing the RTP. Under Asterisk SIP Settings NAT - Yes IP Configuration - Public IP (I We have a recent FreePBX installation, hooked up to an SIP trunk via pjsip. Turn off DMZ and instead forward the UDP port range for RTP (default 10000 to 20000) and the Bind Port (your value 5185) to the PBX. 1p CoS SIP: 4 802. All extention work on pjsip. png 1174×210 11. Short and quick story, incoming calls are rejected with SIP/2. Please provide the full log with protocol debugging on (e. Laboratory experiment: I have a clean install of Freepbx 16. 5 min. c:24122 handle_response_peerpoke: Peer ‘ovh’ is now My phone and freepbx are on the same LAN (192. 25. 53 Asterisk: 16. c:4175 retrans_pkt: Retransmission timeout reached on transmission 4da5e510d704f4515dc293959e416 for seqno 1 (Critical Response) – FreePBX Community Forums Chiamata da smartphone. there is no audio on the call and the hangup doesn’t work, if left long enough it will timeout. 1p CoS RTP video: 6 802. xxx (S) 255. Only affecting Polycom SoundPoint IP 321 Phones. Can you rip_timeout_hold. Hoping the outbound calls won’t be longer than 1 hour, this solved the problem I was having. Currently, if voicemail is not enabled then the Ring Time (or the system-wide ringtimeout) is ignored and an "infinite" Dial is issued. If that’s the case, you can confirm that with BYE from the remote end: Make a test call to your mobile. 1 255. When the remote user places a call to me, upon pickup the call will drop in 2 seconds. Our setup Hi I have an account with voipfone and I want to connect my home FreePBX to it. PhonerLite) is actually torn down properly with an explicit hangup notification, but when network connections drop, we can’t seem to get the RTP stream on the trunk side to either time out or otherwise I am having issues where all calls inbound or out will timeout at 30 minutes almost on the button. After establishing a trunk, and specify outbound plan (and inbound, doesn’t work either), and registering a SIP phone (Nortel 1535), I can’t dial out. Our IVR is set to call a Ring Group (a simple memoryhunt on 3 extensions) on timeout. However, I still do not have any audio on the call. 8 KB. Thank you! Hello, I use asterisk 18 with freepbx 15, I configured a phone to use TLS and SRTP. My VOIP Trunk provider (voiptalk. 255 4569 (T) (E) OK (8 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] When I try to call an extension of pbx-a from pbx-b the extension r FreePBX. Flow - Softphone -> Internet -> PBX on Azure -> Provider While making an outbound call, Invite messages (with SDP) are acknowledged by the PBX with 200OK(SDP), the the provider end is sending RTP requests, but FreePBX doesn’t send any RTP. 9631[2021-02-25 19:28:46] NOTICE[2455] res_pjsip_sdp_rtp. On the Adtran, the command is “ip policy-timeout udp 5060 Hi, I’m quite new to this FreePBX/Asterisk, trying to set up a SIP call using a softphone on my Laptop. 8-2208. RTP Timeout: 30. Adjusted RTP Timeout RTP Hold Timeout CHANSIP set for NAT:yes, Static IP Adjusted registration max and min expiry My Freepbx install (fresh install as of yesterday) resides on the lan, and is dual homed, with it’s second NIC handling the SIP phones (Cisco SPA 5xx). _line=yes media_encryption=no timers=yes timers_min_se=90 media_encryption_optimistic=no refer_blind_progress=yes rtp_timeout=30 rtp_timeout_hold=300 rtp_keepalive=0 send_pai=yes rtp_symmetric=yes rewrite_contact=yes force I have a SIP trunk set up with Twilio for outbound calls. My extensions and trunks use PJSIP. 220), and RTP messages appear to be sent between the two endpoints. This are my configuration: Nat config Maybe one charitable soul (or many) can help me out. Problem only happens on calls coming into the system. Now comes the problem. Looking at the logfiles, i see these two errors: “Connected line update prevented” “Retransmission timeout” I opened up the RTP It’s not seeing the RTP stream, so the keep alive kicks in, and tries to insert it’s own “comfort noise” which disrupts the actual RTP. The user dial *72 to forward his calls to his cell phone. To place a call I dial 9-NXXXXX as outbound route. Unfortunately I cannot receive calls. (basically it was installed from your ISO : SHMZ release 6. 21/32 via 10. guenni (guenni) July 31, 2022, 9:43pm 1. 3. rtp_timeout_hold : 0 sdp_owner : - sdp_session : Asterisk security_mechanisms : security_negotiation : no send_aoc I have set “RTP Timeout” and “RTP Hold Timeout” to 3600, now the calls seem to last longer than 30 seconds. More likely than not, a call that times out does not ring back to the parking device (even though I have that option on) and does not ring through the alternate destination (set to a queue). I’m actually connecting to an extension on their Virtual PBX. Setup another external extension but this time, the setup is using sip on TLS port Hello! I ran into a problem that occurs after restoring from a backup. Weird thing is it works fine if you terminate the transfer in < 3 min, but after 3 minutes it just sticks and does not terminate the call until the rtp timeout?? So confused on this. beaglebone-asterisk. i even created one extension. c: Disconnecting call ‘SIP/aaa-00000003’ for lack of RTP activity in 31 seconds . For the call drop issue, the usual problem is a NAT association timeout. c: Channel Hello, I’m having line issues with one of my remote users. 23. Is this the default setting? Is this the best setting to use? Is it possibile this usign Freepbx ? Thanks. 8. conf:rtp_timeout=30 pjsip. When I dial out to that same remote user, the line doesn’t ring, it just goes to voicemail. c:29644 check_rtp_timeout: Disconnecting call ‘SIP/NTC-00000063’ for lack of RTP activity in 31 seconds. FreePBX. Things like “core set debug 3” and “core set verbose 7” and “pjsip set logger on” and “rtp set debug on” are your friends there but the CLI offers tab-completion, so type in the first word or two then hit Tab to see all the options, which vary based on your Asterisk version. PEER Details: host=<<>> type=peer context=from-trunk disallow=all allow=ulaw,alaw qualify=yes qualifyfreq=10 Have I missed something? FreePBX. system (system) Closed June 4, 2022, 7:38pm FreePBX 12. We have these 30 seconds Hello, We’re trying to set up an Asterisk server with SIP-over-TLS and SRTP capabilities, and after using AsteriskNOW! for a more standard (UDP/RTP) setup, we decided on the FreePBX distro, as it seemed to meet our needs. After I attempt a transfer I still send and receive audio on the call, however after about 30 seconds I run into the Request Timeout Issue and All of our transfered calls get stuck in this weird limbo stage and eventually timeout from rtp after a hangup. our setup: free pbx 2. The default For a standard setup with a FreePBX/Asterisk PBX onsite, you will need the following on the Sonicwall: A Port Forwarding rule of 10000-19999-UDP for the incoming RTP - sometimes you can get away without this rule - depends on the ITSP - Put it in anyway. 12. STUN: makes no difference if using a STUN server or not, for this issue for me. c:29968 check_rtp_timeout: Disconnecting call ‘SIP/out_01212120992-0000003e’ for lack of RTP activity in 31 seconds. Running Asterisk 1. 19 with Asterisk 15. Why not do it in the GUI? Settings → Asterisk SIP Settings [root@freepbx asterisk]# grep rtp_time * -m2 pjsip. No Audio 2. I tried with devices of different manufacturers and with a softphone, whit the same result. In the SIP General settings, RTP REGISTER should only apply to new calls in the reverse direction to the registration. Here is my trunk config: ;allow=g729 ;uncomment this line if you have G. I have multiple other FreePBX switches in production that have never had this problem before. 24 and for my own reasons I changed PJSIP ports to 5080 and I have UDP & TCP enabled. conf [123456] type=aor contact=sip:mydomain. sip show peers 100/100 XX. Local RTP port: This parameter defines the local RTP-RTCP port pair the HT-386 will listen and transmit. 5060, 5160 and 10000-20000 ports are forwarded from the public ip Cannot make calls between extensions on freepbx. 37 with asterisk 16. This will result to one way audio or call drops after 30 seconds. some more random information - A cable company is my ISP. I can get it working by hardcoding the endpoint in pjsip_custom. After 32 seconds the call is disconnected. 24 / Asterisk 16. I have several FreePBX instances and all behind Sonicwalls. Thanks for help. 1804 (Core), working on Asterisk Asterisk 14. We have recently been getting call timeouts or one way silence on calls and I have no idea where to look: This is a call log from when it timed out and hung up: Please let me know if I need to provide any information [2015-08-27 09:42:02] VERBOSE[26102][C-0000001b] bridge_channel. The problem occurs about once a week. But when sip client holds the call this option is not works correctly. 255. 0) with PJSIP and running into a problem when my endpoint disconnects form the network while the call is in progress. My scenario is this: I have two rtp_timeout : 0 rtp_timeout_hold : 0 sdp_owner : - sdp_session : Asterisk send_diversion : true I can also confirm that the packet trace on the FreePBX server can see both RTP streams coming into the server. I am using FreePBX Distro 14. If everything is configured correctly, a long phone call (like the 2 1/2 One of the common issues in FreePBX is the Lack of RTP Activity. c:29402 [2017-06-13 04:42:21] NOTICE[1766]: chan_sip. So it seems I can accomplish TLS/SRTP from server to provider. What works: • Zoiper phone inside the Lan same LAN the freepbx box is on • Sip Trunk to Tynlex • Inbound/outbound calls to Zoiper phone on the Lan • I set under sip settings RTP Timeout and RTP HOLD TImeout to 60 seconds. Applications / Modules I misunderstood. 169, Asterisk 13. This bit of gui We are having an issue with calls timing out after 30 seconds. 55. A few seconds after answering, hang up the mobile and How can you change the time-out in order to eliminate those that have fallen? FreePBX Community Forums Timeout to 32 seconds. Network Settings: SIP address remapping: Enabled using externaddr Externhost: Externaddr: 96. I am having RTP issues with my I am using Freepbx with Asterisk version 13 for connecting to the SIP Trunk. I, too, see calls Right now I am running dual nics on my Freepbx box. 1) what brand and model modem are you using? Please answer this question. I can make an outgoing call from X-Lite. From this VLAN, it connects to local and SSL-VPN clients, in their own respective VLANs, and to the trunk, on the 2nd WAN interface of the firewall, which is the sole routing device. I found different configurations on the internet, and I started with the first two servers, both phones can ring but no voice. c:29987 check_rtp_timeout: Disconnecting call ‘SIP/Airtel_SIP-00000043’ for lack of RTP activity in 31 seconds. I chose a random number of 90210 and populated the GUI parking lot timeout field with it, then queried the parking lot from the asterisk CLI and get: but you may run into limits with RTP timeouts in Hello, I have setup my Inbound Routes and Time Conditions if I set my Time Conditions to a ring a ring group everything is fine and the phones rings as expected when I call my number. 0 mwi_subscription namedcallgroup namedpickupgroup outbound_proxy qualifyfreq rewrite_contact rtcp_mux rtp_symmetric rtp_timeout rtp_timeout_hold secret send_connected_line sendrpid sipdriver timers timers_min_se transport trustrpid user_eq_phone webrtc minimum_expiration refer_blind_progress Hello, I call between two endpoints, 23 (IP: 192. What happens is that the call hangs up after Hello all, Good day, I am new on Freepbx config’s. Yes that’s a For the call drop issue, the usual problem is a NAT association timeout. increasing UDP Other from 30 seconds to 300), and nothing seems to work. A The ‘30 minute’ part screams NAT session drop, and is often a symptom of an incomplete RTP path. I guess the RTP Timeout (which was set to 30) caused this issue. 65-26 and asterisk 11. When given the time and extension looked in /var/log/asterisk/full log and see the following entry which was one of the reported dropped calls: [2020-02-12 11:28:36] NOTICE[5602] chan_sip. I have looked and looked for a solution over the last week and still cant find a resolution. [2506]: chan_sip. The freePBX box need to know whether you need to NAT from public IP addresses to public internal addresses since it doesn’t know if you’ve configured the ethernet inteface of the freePBX box on a public ip address directly (really bad idea BTW) or on a private internal address that is having a public IP address forwarded to it (like a VIP Good morning, I have a serious problem, the context is: the FreePBX device is connected to a Mikrotik ruoter, he registers to an external voip provider correctly. RTP Timeout: 0 (Disabled) RTP Hold Timeout Hello All, My asterisk is behind firewall. I’m aware that 30 seconds is the RTP timeout, but changing it to 45 seconds doesn’t appear to make the drops happen at 45 seconds, they still drop at 30 seconds. There are two different Internet lines on the Mikrotik router. While doing some troubleshooting for this primary issues, i went to login to look at my advanced sip settings to check my nat s Hello. The call does not hangup or disconnect, the audio quits working. I’m just giving a taste of my office installation and Freepbx 15. Hi, I have the following problem to solve: chan_sip. aor_custom_post. I find this written in LogReport. sng7 I am trying to connect my Cisco IAD2432 Version 15. My Freepbx is in DMZ and there is no other device that uses 5000-6000 range port. Behind a NAT Firewall, Using a non Go to the SIP settings under media and RTP settings and change the rtp timeout value to be higher if you're using a FreePBX based version. FreePBX is in the office. 5-1807 i7, 16GB, SSD drive System will make an outgoing phone call with no issues, incoming calls drop AUDIO after 15-15. I have this set in my “Asterisk SIP Settings”, RTP Port Ranges. I’m running and asteriskNOW(1. That i ran into a couple of problems one would be a oneway audio If i call from my office phone to my cellphone meaning if i talk on Good morning all! I have a PBX that has been running flawlessly for over a month. Hi to all, I’ve successfully connected 2 pbx each other by IAX2 trunks: freepbx*CLI> iax2 show peers Name/Username Host Mask Port Status Description System2/System1 151. 35 for seqno 23139 (Critical Respons I upgraded my installation of FreePBX to version 17 and I am running into an issue with the FXS ports on my Cisco SPA8800. When you made the change was it only for your SIP INVITES? If so you would need to include this [2015-06-25 22:01:24] NOTICE[1911]: chan_sip. 8 and FreePBX 2. I have got an issue where our calls gets Hung in between for couple of seconds. Not outbound calls. A new connection will be created, and The calls may be ending because of an RTP timeout, make sure the sip settings in the FreePBX GUI have the right subnets and/or public ip. Looks like you better dig through those traces! FYI, you can do a "sip set debug peer broadvox-tx" in Asterisk to get SIP trace for a particular peer without dealing with tcpdump. Hy there, does anyone has an idea weather my phone can not register at extention 001 but can register at extention 002? rtp_timeout=30 rtp_timeout_hold=300 send_pai=yes rtp_symmetric=yes rewrite_contact=yes force_rport=yes language=de_DE I had configured my FreePBX with some extensions (SIP trunk) and everything worked fine, but some day my public IP had changed, and all extension were unable to log in. I tried changing different settings unsuccessfully. I am trying to figure out syntax in the pjsip. rtp_timeout : 0 rtp_timeout_hold : 0 sdp_owner : - sdp_session I’ve spent hours on this. Now, I want for the Asterisk to always stay in between the extensions, hence, no directmedia, no reinvite, etc. com After reading a gazillion posts I have gotten the inbound calls to stop dropping and sound clear. Phones are Grandstream GXP2XXX with latest firmware. I connect a phone to it using the pjsip protocol. RTP hold timeout: 300. 11 with latest module upgrades + asterisk 13. Further investigation shows that it’s because of no codecs being negotiated Capabilities: us If you are not too far into this install you would be better off dropping AsteriskNow and loading a real FreePBX distro. com [123456] type=endpoint aors=123456 tos_audio=ef tos_video=af41 cos_audio=5 cos_video=4 allow=g722,ulaw,alaw,gsm,g726 context=from it’s freepbx 15. In zulu it shows the call allowing to receive the call, but these calls have no sound and after 30 seconds they are disconnected. What is the preferred method to adjust the rtp port range? Thanks in advance. 11 with Asterisk 11 and need to set the rtpstart and rtpend vaules. On Asterisk logfile i get this Hi there, I have an asterisk/freepbx server locally running (with own dns, but only accessable from lan), which isl only used to call internal extensions, therefore it’s not possible to make outbound calls, as well no need. Home ; Categories ; I'd like to suggest a change in the default dialing behavior of freepbx. The server has a public IP and is not nated. 6. 0/24) The freepbx deetect wan IP address correctly to 90. I suspect it has something to do with the fact there is no MOH so the line is silent but the RTP hold timeout is set to 500. 29. 23). I reall y couldn’t find a solution. XX. When I answer one of those “zombie” calls, there’s just silence. I used to clone a new FreePBX installation from a previous backup, now I can make calls but there is no audio and call drops automatically after 6-7 seconds. 168. However, our (apparently correctly) configured FreePBX 2. Not sure where is the problem, weather Network side or in an config in PBX. General Help. 44. dicko (dicko) September 10, 2013, 5:39pm 18. Cloud Hosted FreePBX (Public IP) FreePBX: 15. When using both ring groups (at 10 seconds timeout each) it doesn’t work. 22. david55 The only part of the FreePBX that might cause something like that is fail2ban, if the address is also generating security alerts I'm new with freePBX. eth0 is 10. We use 80 Zulu Shoftphone. The SIP protocol requires that certain timeout periods are set, within which a response or acknowledgement message must arrive from the far end. Hello! I faced the issue that call drops after 33 seconds due to lack of audio RTP activity. I call from 1153(WebRTC, JsSIP) to 1154(Mobile, Linphone) extension: 1. Scenario: Caller dials DID >> FreePBX answers >> Presses 2 on IVR >> DISA Answers >> PIN Entered >> Dial Tone Active >> Caller dials number via DID to call out >> Call hangs up. FreePBX Community Forums Merge 2 calls. RTP Checksums: Yes. 0731) At first the RTP stream is lost for > 30 seconds the call was disconnected. , to prevent the distribution of viruses The dial command is being given a 300 second timeout by FreePBX. : false rtp_keepalive : 0 rtp_symmetric : true rtp_timeout : 30 rtp_timeout_hold : 300 sdp_owner : - sdp_session : Asterisk send_connected_line : yes send_diversion Hello, I have set up a test FreePBX with the first two trunks with SIP provider Alpha simulating a “failed” state and the third trunk with SIP provider Beta is still active. Try setting RTP keep alive to 1 in Asterisk SIP Settings (it’s 0 by default) and make sure that on your firewall the RTP ports are forwarded to your PBX. (Must be even). 1 dev eth0 Hi! I apologize if this topic has already been discussed before. 5 (Final)) Issue we are facing now is that dial timeout for extensions which are in the ring group being set by dialparties. conf the option canreinvite = no, in advanced settings the NAT = no or yes, I So it’s not happening at exactly 30 seconds like some other very common issues related to local network address/RTP ports forward RTP ports on PBX set to 10001-20000. I am having RTP issues with my Freepbx installation. 40, and restore a backup on it via Hello, Currently I am using Freepbx 15. Providers. Thank you so much. 0 I have recently switched over and started using a different provider (Voyant -> Twilio) because they are discontinuing their SIP trunking services. I am having issues with call reliability with a MicroSip Softphone on a PC. 1 with FreePBX 2. Strict RTP: Yes. I just installed the latest version of the Distro and upgraded all of the modules. 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. I give up and need your help. The file says that ";rtp settings are defined in the chan_motif freepbx module" I couldn’t find anywhere in the GUI to set these vaules. 5. 1 using chan_pjsip, this use to work find on Asterisk 13 using chan_sip, but I am unable to get it fully working on chan_pjsip. I set my ip as local (Trusted ) But there is still no sound, but there is in local area. The trunk SIP with OVH is OK. rtp_timeout=30 [0000002](+) rtp_timeout=30. Twilio-FreePBX and then my test device is the simple X-Lite from CounterPath. I run a UniFi USG, I’ve played with the UDP timeout settings (e. from my laptop, i can be able to ping the freepbx server on cloud without any packet loss. My provider has stated it is definitely being disconnected at the PBX i have taken music on hold off all trunks and groups i can find. NOTICE[1670]: chan_sip. Hi - I’m having an issue were Zoiper softphones on the internet aren’t communicating in to a PBX on my LAN. I'm not familiar with Freepbx. rtp_timeout is set in Asterisk SIP Settings and afaik, it’s a single I’m using the latest beta 2. 11 is being unstable in maintaining TLS connections. [2018-02-15 16:16:29] NOTICE[6591]: chan_sip. 1 sip trunk eth1 is 10. c:29611 check_rtp_timeout: Disconnecting call 'SIP/YYYYYYYYY-00000003' for lack of RTP act Hello everyone! For three weeks we have installed the latest frepbx distribution. 2) What brand and model router are you using? Edit: I see ubiquiti usg3 in your profile. The phones that are connecting are nated and are not working properly. I am attaching the information attained using pjsip set When I have the console pulled up using asterisk -rv I can see the RTP setups such as: > 0x7faebc1b0620 -- Strict RTP qualifying stream type: audio > 0x7faebc1b0620 -- Strict RTP switching source address to WANIP1:62654 > 0x7faec41951f0 -- Strict RTP learning complete - Locking on source address PROVIDERIP:25966 > 0x7faebc1b0620 -- Strict RTP The system is Asterisk 1. This is also a very time RTP Timeout: 30 RTP Hold Timeout: 3600 RTP Keep Alive: 60. c:26488 check_rtp_timeout: Disconnecting call ‘SIP/100-0000005b’ for lack of RTP activity in 31 seconds. 1p CoS RTP audio: 5 802. RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled Latest distro, Asterisk 11 I am registered at Flowroute and sip peer phones show up, I re-booted router, now IP Trunks Online shows 0. The Call Forward Ring Time is set at 30 seconds (default), so in theory, the cell phone should ring for about 30 seconds and the call should then be redirected to the voicemail on the FreePBX. If UDP Unreplied timeout is, for example, 17, and the NAT Keep Alive Interval is 20, then the corrupted connection will timeout or close. This topic was automatically closed 7 days after the last reply. The problem occurs both on a machine behind a draytek router with DMZ on the PBX and on a machine in the cloud when without any nat. When I do sip show peers it is showing their internal IP and not the public IP as it should be showing. All the extensions in the office is working well. In the log files i can read this: NOTICE[2954] chan_sip. [5152]: chan_sip. It says Host Unspecified, I am not sure what or where to set that, it was all working fine before I rebooted the router, I did not change anything in Freepbx. This is likely a typo, it should be: rtp_timeout_hold Either way. org /downloads/ ) I've been playing around with it, getting sip I have a few extensions that I want to authenticate using their IP address instead of a password. Have a new rash of one way audio problems. They’re actually pointing to a host that doesn’t exist, but that could also occur if our SIP provider’s SIP switch or the ISPs to their equipment were down, their datacenter, etc. g. Show your Flowroute trunk config so we can make sure it’s not messed up. And I tried to change the value of RTO time out In settings -----> Asterisk SIP Settings Such as the following, but no effect : RTP. conf but that is auto-generated. Italiano. they are landing in an asterisk instance and get forwarded to another asterisk instance. system (system) Closed July 5, 2019, 4:26pm 13. Calls into the extension or out of it send and receive audio fine. rtp_timeout_hold : 0 sdp_owner : - sdp_session : Asterisk send_diversion : true send_pai : true send_rpid : false When connecting to FreePBX via PJSIP from outside the local network everything works just fine most of the time. These are my settings for a Setup working with normal external extension. This resulted in a page that lasted an untold amount of time (1 hour+) that effectively locked up the paging system. when I run the command: asterisk*CLI> pjsip show endpoints Endpoint: 300/300 Not in use 0 of inf OutAuth: 300-auth/300 InAuth: 300-auth/300 Aor: 300 1 Endpoint: 301/301 Not in use 0 of inf OutAuth: 301-auth/301 InAuth: 301-auth/301 Interesting findings there. 0 488 Not acceptable here. It is the base RTP port for channel 0. com trunk. All I am missing is a route somewhere I feel. I'm running FreePBX 13. Which port am I missing ? Thanks Aurélien I do not understand this entry or recognise the ip address: [2014-11-06 11:14:13] WARNING[2463] chan_sip. 000 \ 20. I was able to configure everything so that my 200 phones (snom D715) would work fine. 0 (we’ve tried 13 and 14 also) after 8 concurrent incoming calls placed in a queue the trunk looses the registration and (obviously drops all the calls). 1p CoS RTP text: 5 Jitterbuffer enabled: No. Is there any way to set a timeout on the length of pages? We had a user page then leave the phone off-hook. 22 255. RTP Timeout : 90 RTP Hold That is why the FreePBX developers exposed those setting in the SIP Settings module so you can configure for your deployment. 197. agi to 2 seconds by some reason For example, we have an extension 5000 (in ring This is an interesting topic for me as well. Domgi (Domenico) August 15, 2021, 4:40pm Hello. c:29402 check_rtp_timeout: Disconnecting call ‘SIP/600-00000009’ for lack of RTP activity in 31 seconds No RTCP candidates; skipping ICE checklist (0x7f6db831fbe8)” warning appeared and hangup . Everything internal works like a charm. For the RTP packets Im not able to hear calls in another network it was working fine but now i just received the calls but im not able to hear them although they are able to hear me. 17. 254. The 5060 port is open on my firewall and modem. I have an issue with calls on hold disconnecting after 90 seconds. 74(11. including SIP trace, at pastbin. After a lot of research and debugging I’ve tracked the problem down to the REGISTER packet that is Hello I am testing all i can seem a few days and i can not conect my freepbx to OVH provider. 18. configuration, freepbx. How do I fix that? Thank you. (asterisk 1. c: Disconnecting call ‘SIP/227-00004dbe’ for lack of RTP activity in 31 seconds The person complaining about the dropped . hxmfd utucqe ujvlsae bmfcwu xcyrk ntqw pbfbd vrtjp iswtumnt zzyuxmf