The UDP port is as specified in Asterisk SIP Settings.
How many extensions? The only way to do it in bulk, is to use the Bulk Handler, but there are substantial differences in the CSV format between pjsip and chan_sip, I would expect it to be almost as much work as submitting one by one in the GUI unless there are a lot of extensions.
You can’t, there is no method for it. You would have to implement a proxy like Kamailio or OpenSIPS to deal with that as they would let you parse/read/rewrite/etc the SIP requests before they hit the PBX.
Use any softphone that supports remote provisioning and does not permit the password to be viewed in the settings. Then, the end user won’t know the secret for his extension so won’t be able to use another client.
I have a pair of new pbxs both running 15.0.16.44 (full updates as of this writing) with backup and restore in a warm spare configuration.
Backup and Restore are working. However I have found 2 things that do not copy over to the warm spare. I do not know if its a configuration option or an error.
Any of the files edited with ‘config edit’. Specifically my custom dialplan in extensions_custom.conf does not appear anywhere on the warm spare.
Dialed Number Manipulations Rules on Trunks also do not copy to the warm spare.
If anyone has any information on these two items I would be appreciative.
Thanks
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This was the previous topic, I finally got my hands on the adapter to put it at a different site.
So I got the unit back and was able to do some further digging;
So I did notice I was able to find this in the log when attempting calls;
res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“Anonymous” sip:anonymous@localhost’ failed for ‘REMOTEIP:5060’
I tried digging into the credentials and re-creating the extension and seeing if I could find what was causing it but no luck, changed the password to a simple one, updated the Display name and User ID in the SPA configuration to see if that was the issue.
So although the adapter worked when it first went out, something happened that made it stop working even when I brought it back in for testing.
The only “fix” I was able to get it working was to factory reset the unit and I set it up as a new extension and it worked.
I took the unit off site to my place and tested it and had no issues, brought it back and tested it on our separate testing network and it worked fine.
Then it went back out to another remote site (same area as before, just down the street to another location)
BAM same issue, they were able to initially make an outbound call, it showed up in the Asterisk Info section and I was able to see that it’s unavailable however, but it did show the IP address of where it had connected from. This location has a different ISP and doesn’t have any extra router attached to it as the old location had a Google Router.
The IP address is whitelisted in the Connectivity > Firewall section and is not listed in the Sys Admin > Intrusion Detection as being blocked, I have also whitelisted it there too.
What I see in the Asterisk Info Tab:
Endpoint: EXTENSION#/EXTENSION# Unavailable 0 of inf
InAuth: EXTENSION#-auth/EXTENSION#
Aor: EXTENSION# 1
Contact: EXTENSION#/sip:REMOTEIP@REMOTEIP:50 2a732c4636 Unavail 0.000
–
So it did connect and the contact point showed the remote IP
I’ve enabled PJSIP Logging for the host in question per the prior thread and I just see the request being sent out over and over, but no response this just gets sent out over and over
<— Transmitting SIP request (450 bytes) to UDP:REMOTEIP:5060 —>
OPTIONS sip:EXTENSION#@REMOTEIP:5060 SIP/2.0
Via: SIP/2.0/UDP PBXWANIP:5060;rport;branch=z9hG4bKPj32ffdfb7-43b9-4737-9bae-490295396c93
From: <sip:EXTENSION#@PBXLANIP>;tag=7fb51a69-b6f4-443e-84f1-bc70fe08e2d4
To: <sip:EXTENSION#@REMOTEIP>
Contact: <sip:EXTENSION#@PBXWANIP:5060>
Call-ID: 7f438b7a-390b-4acd-a9a7-ff4739d0b2b1
CSeq: 57694 OPTIONS
Max-Forwards: 70
User-Agent: PBXact-14.0.13.26(13.29.2)
Content-Length: 0
I have other SPAs to this same box that work fine, this one also worked fine again upon testing here and at my place but as soon as it went back out issue came back.
Different ISPs, different routers, however they are both on an “island” but I would assume these ISPs are using different routes to get to the mainland.
So I’m hoping something I’ve posted might give even the slightest bit further of a clue of what I can do to troubleshoot.
Also it would seem the PBX knows there’s some type of attempt because I would assume the box wouldn’t continuously Transmit a SIP request to that sites IP address randomly? It must know there’s a device trying to register and is transmitting to it but not receiving back if I’m thinking correctly?
I know where it is specified, but this was a 100% clean FreePBX 15 install. One assumes this was set to 5160 prior to restore.
I guess I can poke around the backup and see if they actually are using a non-standard port on the live system (I do not have access to that at the moment).
I went from 13 to 15 the other day. The SIP port bindings of the original system were restored.
They are in mysql, tables kvstore_Sipsettings and sipsettings.
Does a reboot of the phone get things working again temporarily? If so, you may want to check the registration interval/expiration on the phones and try a lower setting, like 2 minutes. Also, are the ip’s of the phones changing at all? It would just help to know in case it’s some kind of firewall issue.
When you say they are failing to register, what are you seeing? Do you see incoming registration requests that are not going through the full handshake? Or do you stop seeing the requests on the pbx? If you’re not familiar with viewing sip packets, you can do this from the asterisk command line with ‘pjsip set logger host ip.of.problem.phone’(assuming it’s a pjsip extension), then reboot the phone or wait until the packets come in. Another option is to run ‘sngrep’ from the system command line and watch for packets that way. You can take a look at the transaction that happens with a successful registration, and compare it with a failing one, assuming you get any packets at all.
Hey, Thank you so much for your help. I got it working now on my virtual box using the below process… but my production one still saying all “circuits are busy now” …
Tried verifying all settings btw my old asterisk and the new, difference was in Asterisk sip setting --> Sip Legacy settings[chan_sip] – > at the bottom in other sip settings: original config was:
match_auth_username = yes
but new one is -
match_auth_username = yes
sendrpid = yes
trustrpid = no
disallowed_methods = update.
so i deleted - sendrpid, trustrpid & disallowed_methods .
now i am able to make any international calls on pbx installed on virtualbox.
Here is the new log from my production server:
https://pastebin.freepbx.org/view/7fd16441
production server is still failing international call… saying circuit busy.
also how do i disable ICE support?? i will looks for where i have option to disable ICE.
Hello,
My goal is to change the ringtone and contact picture for when an emergency after hours calls come in to something funny . There’s a NOC ring group that rings my cell and my Yealink handset.
I set the “Change External CID Configuration” in the ring group to a Fixed CID Value of NXXXXXX. When I dial to an after-hours-test Inbound Route that points to the NOC ring group my cell phone gets shows NXXXXXX (our setup: after hours calls have always came from NXXXXXX), but my handset shows the person that actually called skipping the Fixed CID Value of the ring group.
Does anyone have any suggestions on how to accomplish this? Or any other suggestions on After Hours best practices. It’s easy if it’s just one DID calling the Yealink.
tl;dr need help changing ringtone on handset for after hours line
I didn’t aware of Superfecta. It doesn’t come by default with RasPBX. After installing this module, I found google contact integration system is available there! But after providing my google id I clicked debug but I didn’t get enough information to set it up!
Debug result:
Executing GoogleContacts
Searching Google Contacts for number: 0171710xxxx
{“error”:{“type”:“Google_Auth_Exception”,“message”:“Could not json decode the token”,“file”:"/var/www/html/admin/modules/superfecta/includes/oauth-google/Google/Auth/OAuth2.php",“line”:183}}
Can you please tell me, what is actually wrong here? Or,
do you have any link of complete setup guide for Superfecta? I found some links on internet but those are very old and not working anymore!
Thank you!
Vitelity has failover IP for IP authenticaton.
The public ip in freepbx can be set to a dynamicdns name. Problem solved?
@dicko I actually find a majority of hacks are actually from tftp. They will request on ever mac until they get a password. Since there is no password, there is no fail2ban. Then once they get the password they are able to connect. Same probably for non password protected http provisioning.
I won’t disagree with that but I don’t use tftp nor http for provisioning. So udp/69 is closed and thus never seen.
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