@syadnom wrote:
Ok, I’ve install the OSS endpoint manager. I can’t find any T3x or T4x phones but I do see their info in
/var/www/html/admin/modules/_ep_phone_modules/endpoint/yealinkv70I just cant find it in the UI.
What am I missing?
Posts: 1
Participants: 1
@rshah010 wrote:
Hi All,
Has anyone had any success installing FreePBX on Azure (not AWS)? I found some old threads from 2013 - 2016 with scattered info.
I was able to install it using the instructions in the wiki, but the lack of commercial modules like Endpoint Manager isn’t great.
I’m hoping someone here has had better success and can give me some guidance.
Thanks!
Posts: 1
Participants: 1
@mike_b wrote:
I am having some trouble with my Asterisk/FreePBX system as it seems to drop the connection to the SIP provider. I found out that the COMCAST DNS server seems to be pretty unreliable, so I added a couple of other DNS servers, and things seemed to be improving, but then it dropped out again (although it reconnected after an hour or so (around 3 am) by itself. I started to look at other possibilities and stunbled upon the fact that port 5060 is used for signaling. I checked my firewall and 5060 TCP was forwarded to my system, but not UDP. So I set up another rule to forward UDP5060 to my system. Almost immediately I started to receive what I think are hacking attempts (see below). First I tried to block the IPs through my firewall, but that did not seem to work. Even though the firewall says it was blocking the IPs, the hacking continued (from several IP addresses). I then disabled the UDP5060 again, but the attempts continue. A reboot of the system did not do anyhting either.
Is that just coincidence, that the hacking attempts started when I forwarded 5060 UDP? If not, why havent they stopped after I disabled the UDP forwarding again?
Here is a typical log entry:
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Executing [99901500420556674626@from-sip-external:1] NoOp(“SIP/x.x.x.x-0000000a”, “Received incoming SIP connection from unknown peer to 99901500420556674626”) in new stack
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Executing [99901500420556674626@from-sip-external:2] Set(“SIP/x.x.x.x-0000000a”, “DID=99901500420556674626”) in new stack
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Executing [99901500420556674626@from-sip-external:3] Goto(“SIP/x.x.x.x-0000000a”, “s,1”) in new stack
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Goto (from-sip-external,s,1)
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Executing [s@from-sip-external:1] GotoIf(“SIP/x.x.x.x-0000000a”, “0?checklang:noanonymous”) in new stack
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Goto (from-sip-external,s,5)
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Executing [s@from-sip-external:5] Set(“SIP/x.x.x.x-0000000a”, “TIMEOUT(absolute)=15”) in new stack
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] func_timeout.c: – Channel will hangup at 2018-04-05 16:05:25.424 MDT.
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Executing [s@from-sip-external:6] Log(“SIP/x.x.x.x-0000000a”, "WARNING,“Rejecting unknown SIP connection from 185.107.80.8"”) in new stack
[2018-04-05 16:05:10] WARNING[3490][C-00000008] Ext. s: “Rejecting unknown SIP connection from 185.107.80.8”
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Executing [s@from-sip-external:7] Answer(“SIP/x.x.x.x-0000000a”, “”) in new stack
[2018-04-05 16:05:10] VERBOSE[3490][C-00000008] pbx.c: – Executing [s@from-sip-external:8] Wait(“SIP/x.x.x.x-0000000a”, “2”) in new stack
[2018-04-05 16:05:12] VERBOSE[3490][C-00000008] pbx.c: – Executing [s@from-sip-external:9] Playback(“SIP/x.x.x.x-0000000a”, “ss-noservice”) in new stack
[2018-04-05 16:05:12] VERBOSE[3490][C-00000008] file.c: – <SIP/x.x.x.x-0000000a> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[2018-04-05 16:05:18] VERBOSE[3490][C-00000008] pbx.c: – Executing [s@from-sip-external:10] PlayTones(“SIP/x.x.x.x-0000000a”, “congestion”) in new stack
[2018-04-05 16:05:18] VERBOSE[3490][C-00000008] pbx.c: – Executing [s@from-sip-external:11] Congestion(“SIP/x.x.x.x-0000000a”, “5”) in new stack
[2018-04-05 16:05:23] VERBOSE[3490][C-00000008] pbx.c: == Spawn extension (from-sip-external, s, 11) exited non-zero on ‘SIP/x.x.x.x-0000000a’
[2018-04-05 16:05:23] VERBOSE[3490][C-00000008] pbx.c: – Executing [h@from-sip-external:1] Hangup(“SIP/x.x.x.x-0000000a”, “”) in new stack
[2018-04-05 16:05:23] VERBOSE[3490][C-00000008] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/x.x.x.x-0000000a’
[2018-04-05 16:05:42] WARNING[2245] chan_sip.c: Retransmission timeout reached on transmission 5419fc3efc9767227533005d22a8c9a2 for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Posts: 5
Participants: 3
@tbrpc wrote:
Tried to see if this topic was covered - no luck. Ditching Elastix (which has FreePBX 2.11.04 in it). Started following instructions for Conversion Tool and realized it assumes you are moving to new machine. I would like it on existing machine (an appliance with a Sangoma PSTN card). Do I just go through the motions in instructions with a new box (spare PC) and when complete, cpio or tar the whole thing over to existing original (donor)appliance? Part of procedure has the new machine registering (activated), but ultimately I would like the donor to be the activated machine with new FreePBX.
Posts: 2
Participants: 2
@avayax wrote:
Had to restart Asterisk via fwconsole restart and got this error.
Anybody have seen this before?The command “runuser ‘asterisk’ -s ‘/bin/bash’ -c ‘cd /drbd/httpd/www/html/admin /modules/ucpnode/node && mkdir -p /home/asterisk/.pm2 && mkdir -p /drbd/httpd/ww w/html/admin/modules/ucpnode/node/logs && export HOME=/home/asterisk && export P M2_HOME=/home/asterisk/.pm2 && export ASTLOGDIR=/var/log/asterisk && export ASTV ARLIBDIR=/var/lib/asterisk && export PATH=$HOME/.node/bin:$PATH && export NODE_P ATH=$HOME/.node/lib/node_modules:$NODE_PATH && export MANPATH=$HOME/.node/share/ man:$MANPATH && /drbd/httpd/www/html/admin/modules/pm2/node/node_modules/pm2/bin /pm2 start /drbd/httpd/www/html/admin/modules/ucpnode/node/index.js --update-en v --name ‘’‘ucpnode’’’ -e ‘’’/var/log/asterisk/ucpnode_err.log’’’ -o ‘’’/va r/log/asterisk/ucpnode_out.log’’’ --log ‘’’/dev/null’’’ --merge-logs --log-da te-format ‘’‘YYYY-MM-DD HH:mm Z’’’’” failed.
Exit Code: 137(Kill (terminate immediately))Output:
[PM2] Spawning PM2 daemon with pm2_home=/home/asterisk/.pm2
Error Output:
Killed (core dumped)
Posts: 1
Participants: 1
@kendalldever15 wrote:
I have a customer using PBXact and from the AA day one of the options is sales. If no one answers if goes to a sales mailbox that has a unique greeting. The AA night goes to a different mailbox because they have a different greeting. Currently all the sales people have two BLF keys to monitor the two separate mailbox’s.
What they are asking for is to keep the two unique greetings but have only one BLF key monitor both mailboxes.
Is it possible or is there a way to perhaps forward a message and delete it from one of the mailboxes?
Posts: 3
Participants: 2
@chasemixon wrote:
ok the only change we have made is the sip provider, he is also my ISP, so I’m not sure what is going on, but I do have some call traces, so I’m hoping one of you lovely people can tell me where it is going awry.
so my ext is 137, and as you can see a friend of mine calls my number it rings my ext and does a followme to my other phone 714-1505# when I answer the call he can’t hear anything I say and I can’t hear anything he says.
please help!
Posts: 1
Participants: 1
@tbrpc wrote:
I got my new temp SGX 7 distro activated and started the conversion process. All is well on new distro machine…“Waiting for Donor…”. Ran the same Curl command on the Elastix machine and it does nothing. Brings up the next command prompt line. I checked and I have Curl 7.15.5 on it, so what is happening? Apparently nothing!
Posts: 2
Participants: 1
@Craigst wrote:
I wana set up a dedicated raspberry pi 3 for a phone system with zoiper
- i have 2 BT home lines just dont wana have a home phone so is there a SIP way to connect them into they system ( i know it can be down with business easy )
- i have a small business that has 2 lines 1 for driver and other for public , quite busy so need to be able to put on hold and depending on the number that is incoming ( backup system as will be diverted from a control room)
- recommend a SIP as this is a back up / emergency system dont really wana pay unless i use it (in uk only) and on outgoing calls to display the company number
last where to start i just have to server all set just wana connect sip to it then set up acconts and connect to my pc and my personal phone
ANYHELP will be much appricated
Posts: 1
Participants: 1
@johnens wrote:
Just upgraded a system from FreePBX 13 to 14. Since updgraded, the system no longer recognizes the Digium A4B analog card in the analog tab of DAHDI Configuration.
Here’s the output from dahdi_hardware:
pci:0000:01:00.0 wcaxx- d161:8010 Digium A4B
Under Module settings, I’ve only selected wcaxx. Under Modprobe settings, wcaxx is selected for Module Name.
Also tried running dahdi_genconf, and all that was displayed was
Empty configuration – no spans
What else may have changed during the upgrade that would break this functionality?
Posts: 2
Participants: 1
@ptruman wrote:
Lo there
My server recently died and was rebuilt, and I found that my FreePBX backup was anything but complete. Luckily I managed to recover 99% from the previous disk, but I have one small problem left.
When setting up (before the restore) I tried to install the announcements module.
It failed, with an error I couldn’t read (it was bigger than the popup install window and couldn’t be scrolled).I keep being prompted to upgrade it, and get the same error, so I removed it.
If I try to re-add it, it fails again. Everything else is working perfectly.I’ve noticed when I’m running a backup, even with the module removed, I get this message - which I presume is half the problem…clues?
Here is the error
> Initializing Backup 3 > Backup Lock acquired! > Running pre-backup hooks... > Adding items... > Error: Couldn't read status information for table announcement () > mysqldump: Couldn't execute 'show create table `announcement`': Table 'asterisk.announcement' doesn't exist (1146) > Building manifest... > Creating backup...
Posts: 2
Participants: 1
@dillydilly90 wrote:
I want to do some thing that not sure is entirely possible. We’re a small outdoor power equipment dealer with eight employees, four of which who answer the phones. The problem being is that we get slammed during the spring time with phone calls and walk ins. So the employees who answer the phone aren’t always available to answer it and chances are they’ll forget to log in/out of the queue or hit DND.
What I would like to do is roll all the incoming calls (3 channels) into one queue. However I want to have only one agent on the phone dealing with the calls in the queue. Once that agent is finished then the next call in the queue can ring all the phones until someone picks up. That way the others can call out if they need to or assist customers in our store.
I’ve got everything else working like a charm but I not sure how to go about handling the calls in queue.
Posts: 3
Participants: 2
@jtg1991 wrote:
I’ve been trying to figure out the issue with our FreePBX servers for over a month and I’ve come to dead end after dead end. Eventually inbound calls will stop working, however I can still make outbound calls. This used to happen once every two days, now it seems to happen once a day. Trunk provider is VoicePulse:
(obviously I changed some parameters around for security reasons)
type=peer
context=from-trunk
host=*voicepulsehost
username=*username
secret=*password
qualify=4000000
disallow=all
allow=ulaw
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes
rtptimeout=0
rtpholdtimeout=0
rtpkeepalive=20000
session-timers=refuseI initially used context=from-ptsn but still had the issue. I called VoicePulse and they think it’s a registration issue, however debugging the trunk IP shows that packets are going just fine. The office setup has two WAN ports for redundancy, one for each fiber box that connects our pfsense firewall to the outside world. We have two identical FreePBX servers, each one pointing to one of the WANs. For example, Server A points to WAN port A, while Server B is listening to WAN port B. Shutting off one of the servers doesn’t seem to help as the issue pops up again a couple days after. What’ll usually happen is one server will stop taking inbound calls and the other will start listening. For example Server A stops taking inbound calls so Server B starts handling them. Eventually Server A comes back and handles them again. Depending on the day they will flip-flop with one another. Also whichever server is handling inbound calls also handles outbound calls. Just yesterday it got to the point where neither server was handling inbound calls at all, only outbound calls.
Can anyone at all lead me onto the right path? This issue has gotten to the point where I’ve contemplated finding another career because of how bad it’s gotten.
Posts: 10
Participants: 2
@Hawkeye wrote:
Hi, tried to upgrade PM2 and it failed…
Found post from TM1000 and it fixed it…
rm -Rf /home/asterisk/{.npm,.npmrc,.node-gyp,.package_cache}
rm -Rf /var/www/html/admin/modules/pm2/node/node_modules
fwconsole ma install pm2
Posts: 1
Participants: 1
@matthewljensen wrote:
I’m deploying a system that is likely to encounter at its peak around 15 simultaneous calls. It has a symmetric connection of 30mbps (At least that’s what it’s supposed to have, but even at 15 Mbps symmetric, I should be encountering the problems I am), and is the only device on that internet connection. I recently did a test using call files where I initiated a number of call simultaneously. I was on the phone at the time, so that I could hear the deterioration and come up with a viable number of simultaneous calls. Eventually, I figured out that the system could handle about 13, including my call. After that, the quality detorated quickly. From what I’ve read, a normal call takes about 100kbps up and down, so even with a 15mbps connection, I should be able to easily handle 100 simultaneous calls. I ran a speed test and was getting in the 20mbps, even during the simultaneous calls. What could the problem be?
FreePBX is running on a dell poweredge r410, so I doubt that the problem is the processor.
My trunk is Flowroute, which doesn’t have any channel limitations, so I doubt the problem is with them.
But I also think it is unlikely to be solely related to our connection.
Is there anything I can do to do some more tests, or is what I’m experiencing with the call files likely to be similar to what I see in the real world? Can I experiment with codecs? What codecs should I try?
I’m at a loss. Can anyone with some experience weigh in?
Posts: 2
Participants: 2
@robinsonjas wrote:
I am able to register a new deployment via CLI - but is there a way to add the ‘Location’ flag to the command so it doesn’t come in as “New Location” in the Sangoma Portal??
I’m working on some automation for customizing new deployments and would like to add that to the script.
Posts: 1
Participants: 1
@Wasca wrote:
Hi Guys
Running FreePBX 13.0.194.5
Just did an update to XMPP to version 13.0.17.13 and now the XMPP services is not running. I’ve rebooted the PBX and still have the issue.
I can see these entries repeating on the XMPP error log.
2018-04-08 19:18 -04:00: Error: Cannot find module 'lodash' at Function.Module._resolveFilename (module.js:336:15) at Function.Module._load (module.js:278:25) at Module.require (module.js:365:17) at require (module.js:384:17) at Object.<anonymous> (/var/www/html/admin/modules/xmpp/node/node_modules/lets-chat/app.js:11:9) at Module._compile (module.js:460:26) at Object.Module._extensions..js (module.js:478:10) at Module.load (module.js:355:32) at Function.Module._load (module.js:310:12) at Object.<anonymous> (/var/www/html/admin/modules/pm2/node/node_modules/pm2/lib/ProcessContainerFork.js:83:21)Not sure what to do next to fix the missing lodash module.
Posts: 2
Participants: 1
@Solako wrote:
Hello, I have a question. This is leans more towards network configuration.
I have FreePBX 13.0.1942.2 with Asterisk13.12.1 running in a Virtual environment running VMware workstation. The PBX,at time of installation,was connected to a router with a 192.168.1.1 gateway so as to be able to get the downloadable packages. As such, I opted to have the network settings on the VM take a bridged network as those are the configuration settings that the tutorials I was following used. The host machine is in the same subnet with an IP of .77. The Freepbx is .79. when I run ifconfig on Freepbx it show eth 0 192.168.1.79(on the host machine this ia actually ethernet 2.
An incoming SIP trunk has a an IP of 10.x.x.x. It’s terminating on NIC ethernet 1. There sits no router before the IP-PBX. I am wondering how to make FreePBX see the 10.x.x.x. port as the calls to be able to make outgoing calls.
Moreover, may someone could suggest a better method of making this work.
Regards
Posts: 1
Participants: 1
@dominic wrote:
I recently upgraded to FreePBX 14 (on the official distro), and DTMF stopped working (ie I press a key, and the other side doesn’t recognize it). After several hours of debugging, it seems to be related to codec frequency.
I’m using linphone on Android and Windows as my softphones. If I enable a codec in my linphone settings that uses a frequency greater than 8000 Hz (e.g. speex, opus, etc.) even if that codec is not negotiated, DTMF fails. Given that situation, I found this FreeSWITCH bug which I thought might also apply to FreePBX, but as far as I can tell, that’s not what’s going on here.
I enabled rtp and pjsip debugging and ran some tests. I added the logs below.
Both calls wind up using the same codec, but one works and one doesn’t. Obviously, on the successful call we see the DTMF packets are recognized as RFC2833 and they are not on the failed call. I just can’t understand why. Could it be because when it worked the payload number for
telephone-eventwas over 100 and when it didn’t it was under 100? That’s a real shot in the dark, but I am completely stumped.Successful Call (DTMF worked)
Initial INVITE from Caller
[2018-04-06 14:31:55] VERBOSE[6791] res_pjsip_logger.c: <--- Received SIP request (1322 bytes) from TLS:192.168.1.10:36623 ---> INVITE sip:1101@freepbx.server.com SIP/2.0 Via: SIP/2.0/TLS 192.168.1.10:36623;branch=z9hG4bK.x4Iowo7HJ;rport From: <sip:1004@freepbx.server.com>;tag=iQuPf4GBP To: sip:1101@freepbx.server.com CSeq: 20 INVITE Call-ID: Al9P0yUtod Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 702 Contact: <sip:1004@192.168.1.10:36623;transport=tls>;+sip.instance="<urn:uuid:5a3150e3-5e90-4984-b12b-2cfc5ad4a15e>" User-Agent: LinphoneAndroid/3.3.2.2-TG (belle-sip/1.6.3) v=0 o=1004 3003 1159 IN IP4 192.168.1.10 s=Talk c=IN IP4 192.168.1.10 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7076 RTP/SAVP 96 0 8 18 101 a=rtpmap:96 speex/8000 a=fmtp:96 vbr=on a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Yl6b4HKe7GanfgObzlZmCS3TX1iZLmOMW57JJjJ2 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:gkfj8LHaB1PZyV8LLdGVDnMu88YT1FxZPwcgcwL8 a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:kE1BFsEgifzQw9LAyyD80aaIMeFrOlwf1vA4T44PsKS7tlEAHLF552MJ5uGPKg== a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:rXGPaoCCvjC0KuwA0aG3HgmHFCGMd5qGeTfGZKYf2WPxhHWKbRYc7MVlrYOsug== a=rtcp-fb:* ccm tmmbrNotify Caller of Answer
[2018-04-06 14:31:56] VERBOSE[32423] res_pjsip_logger.c: <--- Transmitting SIP response (1040 bytes) to TLS:192.168.1.10:36623 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.1.10:36623;rport=36623;received=192.168.1.10;branch=z9hG4bK.g2CzEVYhu Call-ID: Al9P0yUtod From: <sip:1004@freepbx.server.com>;tag=iQuPf4GBP To: <sip:1101@freepbx.server.com>;tag=30828a9d-cdbf-479a-9016-b48215b31e8c CSeq: 21 INVITE Server: FPBX-14.0.2.10(15.2.2) Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER Contact: <sip:192.168.1.5:5061;transport=TLS> Supported: 100rel, timer, replaces, norefersub P-Asserted-Identity: "User 1" <sip:1101@freepbx.server.com> Content-Type: application/sdp Content-Length: 387 v=0 o=- 3003 1161 IN IP4 192.168.1.5 s=Asterisk c=IN IP4 192.168.1.5 t=0 0 m=audio 12880 RTP/SAVP 0 8 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:k7d54Fzm1HMFS2WU3xr0zyepyrShTeTIqa03aofs a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecvRTP Stream
[2018-04-06 14:31:57] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 004392, ts 565097400, len 000170) [2018-04-06 14:31:57] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000005, ts 565097565, len 000160) [2018-04-06 14:31:57] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 004393, ts 565097560, len 000170) [2018-04-06 14:31:57] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000006, ts 565097725, len 000160) [2018-04-06 14:31:57] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 004394, ts 565097720, len 000170) [2018-04-06 14:31:57] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000007, ts 565097885, len 000160) ..... [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 101, seq 000092, ts 602617351, len 000004) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got RTP RFC2833 from 192.168.1.10:7076 (type 101, seq 000092, ts 602617351, len 000004, mark 1, event 00000008, end 0, duration 00160) [2018-04-06 14:31:59] DEBUG[23733][C-000000a6] res_rtp_asterisk.c: - RTP 2833 Event: 00000008 (len = 4) [2018-04-06 14:31:59] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000110, ts 565114365, len 000160) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 004498, ts 565114360, len 000170) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 101, seq 000093, ts 602617351, len 000004) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got RTP RFC2833 from 192.168.1.10:7076 (type 101, seq 000093, ts 602617351, len 000004, mark 0, event 00000008, end 0, duration 00320) [2018-04-06 14:31:59] DEBUG[23733][C-000000a6] res_rtp_asterisk.c: - RTP 2833 Event: 00000008 (len = 4) [2018-04-06 14:31:59] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000111, ts 565114525, len 000160) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 004499, ts 565114520, len 000170) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 101, seq 000094, ts 602617351, len 000004) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got RTP RFC2833 from 192.168.1.10:7076 (type 101, seq 000094, ts 602617351, len 000004, mark 0, event 00000008, end 0, duration 00480) [2018-04-06 14:31:59] DEBUG[23733][C-000000a6] res_rtp_asterisk.c: - RTP 2833 Event: 00000008 (len = 4) [2018-04-06 14:31:59] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000112, ts 565114685, len 000160) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 004500, ts 565114680, len 000170) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 101, seq 000095, ts 602617351, len 000004) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Got RTP RFC2833 from 192.168.1.10:7076 (type 101, seq 000095, ts 602617351, len 000004, mark 0, event 00000008, end 0, duration 00640) [2018-04-06 14:31:59] DEBUG[23733][C-000000a6] res_rtp_asterisk.c: - RTP 2833 Event: 00000008 (len = 4) [2018-04-06 14:31:59] VERBOSE[23736][C-000000a6] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000113, ts 565114845, len 000160) [2018-04-06 14:31:59] VERBOSE[23733][C-000000a6] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 004501, ts 565114840, len 000170)Failed Call (DTMF didn’t work)
Initial INVITE from Caller
[2018-04-06 14:32:19] VERBOSE[6791] res_pjsip_logger.c: <--- Received SIP request (1406 bytes) from TLS:192.168.1.10:36623 ---> INVITE sip:1101@freepbx.server.com SIP/2.0 Via: SIP/2.0/TLS 192.168.1.10:36623;branch=z9hG4bK.09oNseORt;rport From: <sip:1004@freepbx.server.com>;tag=0bVmil7L7 To: sip:1101@freepbx.server.com CSeq: 20 INVITE Call-ID: giaAPEO5oz Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 786 Contact: <sip:1004@192.168.1.10:36623;transport=tls>;+sip.instance="<urn:uuid:5a3150e3-5e90-4984-b12b-2cfc5ad4a15e>" User-Agent: LinphoneAndroid/3.3.2.2-TG (belle-sip/1.6.3) v=0 o=1004 1441 2486 IN IP4 192.168.1.10 s=Talk c=IN IP4 192.168.1.10 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7076 RTP/SAVP 96 97 0 8 18 101 98 a=rtpmap:96 speex/16000 a=fmtp:96 vbr=on a=rtpmap:97 speex/8000 a=fmtp:97 vbr=on a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/16000 a=rtpmap:98 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SlMxT019CunqOGrAXwXreRhm6kJ9dw8X3Gh6/LhV a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:pWWTCwxlIGi20WcvZCAJC0Jo29dIrJ/9VmDuk0Ig a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:PkYK/oG2F8U53pE6Y/1gu05TqqxgqsfFmEegunUOyMsTpNPA8gAEE7QXULB16Q== a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:US4TIkQ3sl1XK28C7LZPBMGzJBaCGJqxMVt3sg2PBZFIcwv3KomV8ZWZ3DyvRw== a=rtcp-fb:* ccm tmmbrNotify Caller of Answer
[2018-04-06 14:32:21] VERBOSE[30966] res_pjsip_logger.c: <--- Transmitting SIP response (1037 bytes) to TLS:192.168.1.10:36623 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.1.10:36623;rport=36623;received=192.168.1.10;branch=z9hG4bK.MANUueCbB Call-ID: giaAPEO5oz From: <sip:1004@freepbx.server.com>;tag=0bVmil7L7 To: <sip:1101@freepbx.server.com>;tag=1add4390-017a-4856-9973-9946756bba26 CSeq: 21 INVITE Server: FPBX-14.0.2.10(15.2.2) Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER Contact: <sip:192.168.1.5:5061;transport=TLS> Supported: 100rel, timer, replaces, norefersub P-Asserted-Identity: "User 1" <sip:1101@freepbx.server.com> Content-Type: application/sdp Content-Length: 384 v=0 o=- 1441 2488 IN IP4 192.168.1.5 s=Asterisk c=IN IP4 192.168.1.5 t=0 0 m=audio 16570 RTP/SAVP 0 8 18 98 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:oVH9HZ2j56kvQFRcwOdlEi550fdfu7fBr9RhTHRN a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=maxptime:150 a=sendrecv a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16RTP Stream
[2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000004, ts 3208992109, len 000160) [2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019276, ts 3208992104, len 000170) [2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000005, ts 3208992269, len 000160) [2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019277, ts 3208992264, len 000170) [2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000006, ts 3208992429, len 000160) [2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019278, ts 3208992424, len 000170) [2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000007, ts 3208992589, len 000160) [2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019279, ts 3208992584, len 000170) [2018-04-06 14:32:21] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000008, ts 3208992749, len 000160) [2018-04-06 14:32:21] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019280, ts 3208992744, len 000170) ....... [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000078, ts 4113917409, len 000004) [2018-04-06 14:32:23] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000096, ts 3209006829, len 000160) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019368, ts 3209006824, len 000170) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000079, ts 4113917409, len 000004) [2018-04-06 14:32:23] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000097, ts 3209006989, len 000160) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019369, ts 3209006984, len 000170) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000080, ts 4113917409, len 000004) [2018-04-06 14:32:23] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000098, ts 3209007149, len 000160) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019370, ts 3209007144, len 000170) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000081, ts 4113917409, len 000004) [2018-04-06 14:32:23] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000099, ts 3209007309, len 000160) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019371, ts 3209007304, len 000170) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000082, ts 4113917409, len 000004) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000082, ts 4113917409, len 000014) [2018-04-06 14:32:23] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000082, ts 4113917409, len 000014) ........ [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000124, ts 4113924369, len 000004) [2018-04-06 14:32:24] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000139, ts 3209013709, len 000160) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019411, ts 3209013704, len 000170) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000125, ts 4113924369, len 000004) [2018-04-06 14:32:24] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000140, ts 3209013869, len 000160) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019412, ts 3209013864, len 000170) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000126, ts 4113924369, len 000004) [2018-04-06 14:32:24] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000141, ts 3209014029, len 000160) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019413, ts 3209014024, len 000170) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000127, ts 4113924369, len 000004) [2018-04-06 14:32:24] VERBOSE[23851][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.20:48038 (type 00, seq 000142, ts 3209014189, len 000160) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.10:7076 (type 00, seq 019414, ts 3209014184, len 000170) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000128, ts 4113924369, len 000004) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000128, ts 4113924369, len 000014) [2018-04-06 14:32:24] VERBOSE[23850][C-000000a7] res_rtp_asterisk.c: Got RTP packet from 192.168.1.10:7076 (type 98, seq 000128, ts 4113924369, len 000014)
Posts: 1
Participants: 1
@bradm413 wrote:
We have a nightly warm spare backup setup to copy everything over to our secondary server. Recently we had some updates on our primary server, and I’m wondering if major updates could break the warm spare if certain files are copied over via the backup instead of actually running updates on that secondary server.
What is the best practice here? Should I always run updates on both systems? I’d like to have a way to “test” updates to make sure I don’t have any problems in production.
Posts: 5
Participants: 3
