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How to have a backup Inbound SIP, possible?

November 7, 2018, 9:59 pm
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gbaughma is correct. Your DID number is pinned to your SIP provider.

A quality SIP provider has multiple POP’s (Points Of Presence) or Proxies. Depending on how your SIP provider implements failover technology they may handle this on their side or you may have to configure some secondary backup settings.

We use Flowroute and they handle it.

We do however have a second trunk configured on our FreePBX that uses our secondary internet connection to our PBX and office, just in case the primary internet goes offline.

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PJSIP Endpoint Matching

November 7, 2018, 10:03 pm
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Well, I don’t understand what is going on but thought about a possible workaround:

Assuming that calls to/from the extension and outbound calls on the trunk are both ok, try adding a dummy trunk with no registration or authentication and with the OBi IP address in SIP Server (and perhaps also in Match). With luck, after not matching the username it will match the IP address and accept it as onymous.

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PJSIP Endpoint Matching

November 7, 2018, 10:34 pm
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Damn, you’re good, Stewart!

A minimal PJSIP trunk (obi202btx) configured as you described (no Advanced tab settings needed) gets the job done and inbound calls are now routed to from-pstn as expected and ‘Allow SIP Guests’ can be set to No.

Thank you!

@tm1000

Andrew,

Can you shed some light on what’s going on here? Should a bug report be filed with FreePBX or Asterisk?

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PJSIP Endpoint Matching

November 7, 2018, 10:55 pm
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I don’t understand the issue to be honest there’s a lot of back and forth here. What exactly do you think freepbx should do. If it’s change any of the defaults in trunk settings we are against that because they are the asterisk defaults and those are the chan_sip defaults. Hints for settings are also taken from the asterisk wiki.

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PJSIP Endpoint Matching

November 7, 2018, 11:14 pm
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The help for Trunk -> pjsip Settings tab -> General tab says:

Usually, this will be set to ‘Outbound’, which authenticates calls going out, and allows unauthenticated calls in from the other server. If you select ‘None’, all calls from or to the specified SIP Server are unauthenticated. Setting this to ‘None’ may be insecure!

However, this setting determines whether REGISTER’s are authenticated (not INVITE’s). INVITE’s are never authenticated (at least when the Registration option is set to Receive).

When using Registration=Receive, it appears it’s not possible for incoming calls to be matched with the correct endpoint and they always end up going to the anonymous context (from-sip-external). Stewart’s work-around of having a dummy trunk configured for no registration plus manual SIP Server and SIP Server Port settings to collect (match) the incoming call and route it to the desired context is extremely clever, but not a very pretty long-term/permanent fix.

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iLBC codec causes no audio

November 7, 2018, 11:27 pm
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Hi all: I’m trying to use iLBC codec but I found that iLBC has a no audio issue. If both endpoints are using iLBC neither endpoint can hear the other, even though RTP packets are flowing correctly, as checked with rtp debug. Also if an echo test is performed on an endpoint with iLBC, neither the initial explanatory message nor the echo itself can be heard. If one endpoint is using iLBC and the other is using another codec, I tried with ulaw, g729 and g722, iLBC endpoint can’t hear the other non-iLBC endpoint, but non-iLBC endpoint can hear iLBC endpoint.

This has been tested between internal extensions on a local LAN, no NAT involved. This issue only happens when one or both endpoints are using iLBC. There are no audio issues when using any other codec except iLBC.

Asterisk 13.23.1
FreePBX 10.13.66-22

Any help will be much appreciated.

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Disable Intercom Auto Answer

November 8, 2018, 1:04 am
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Actually I’ve been extremly stupid and we can forget about this whole conversation. I’m sorry!

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GUI Issues - Contact Manager

November 8, 2018, 1:30 am
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After updating modules i now get an error, this error occurs when going to the contact manager, also when trying to save any extension. The error i get is;

Whoops \ Exception \ ErrorException (E_ERROR)

Zend OPcache class loading error, class FreePBX\modules\Contactmanager, function install

I’m a little unsure on how to fix this issue.

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Asterisk 13.23 in Repo

November 8, 2018, 3:24 am
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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

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GUI Issues - Contact Manager

November 8, 2018, 3:24 am
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Hi

Please check your modules to not have any Disable module.
fwconsole ma list
fwconsole ma enable xxxxx --> If you have Disable module you can enable from this command

Thanks.

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Need some help to configure SIP trunk with Gateway and POTS lines

November 8, 2018, 3:27 am
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Hello everyone!
There is someone out there that could help me about configuring a SIP trunk in FreePBX in order to connect my FreePBX to a gateway FXO (Yeastar TA410 4FXO ports) ?

my current situations is that i can make inbound calls using only one POTS/FXO port and i’m only able to get INbound calls, if i try to do an outbound i get the “line is busy” message

any advices for the “sip settings” of the trunk?
i would say that

host= the IP of my gateway?
username= usr name that i use to acces to my gateway?
secret=psw that i use to acces to my gateway?
type= ??? by default its “peer” but i cant understand what i should put here! i found in the wiki that i could change it to “user” but it’s not working…

any advices/ideas?

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Internet provider changed - problems with remote extensions

November 8, 2018, 3:33 am
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Hey all,

I’m completely new to voip, but I’m currently the only tech guy at my company, so when anything goes wrong it falls to me.

We have a voip server running FreePBX 14.0.3.2 here in the office, and 10+ phones here on the local network. Mostly Polycom SoundPoint 331s. We also have three employees that work from home, and they each have a Polycom SoundPoint 331 which connect to our server and take part in the whole thing.

Our office changed internet provider a couple days ago, and immediately those remote phones lost connection. I got them back online by having them change the ip addresses of the SIP server and the outbound proxy to point to our new ip address. That makes sense.

However, after the initial jubilation, it became apparent that all calls over these phones are dropping out after about 30 seconds, and now one of them says she isn’t receiving any calls at all. I’ve been reading around, and perhaps the call dropping is due to failing to receive call acknowledgement? But why would the one phone not be receiving calls at all any more?

I’m at my wits end here, especially because I didn’t set this system up and I have a lot on my plate at the moment anyway. So any direction or advice would be welcome!

Many thanks.

edit: I’d love to look at the asterisk log and learn something about this. If I could filter for something in particular, that would be brilliant, but I don’t know what I’m looking for. Any ideas?

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CPU load once in a minute - cron

November 8, 2018, 3:39 am
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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

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Voice IVR - Any new updates or software to use?

November 8, 2018, 3:48 am
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Btw, very cool and informational video! I learned a lot about FreePBX history. Thank you for sharing.

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Dynamic On-Call Destination

November 8, 2018, 4:07 am
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I really don’t understand what you are trying to say here.

Fact: there was a supported way from the GUI to Login/Off from queues when calling externally. This is no longer the case. And I don’t understand why it was removed.
Another fact: We need/want that feature in newer versions.

What difference does it make if it’s a cellphone number, an Extension with followme, or a regular extension?
Again, this is about being able to Login/Off remotely.

No, they need a queue, and again. They need to be able to “Pull the line” (and sometimes login with a few extensions) by calling in.

That’s off topic. We were talking about a supported way of doing that through the GUI.
I know that you can do that with custom context. But again, we are talking about being able to do that in the GUI.

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GUI Issues - Contact Manager

November 8, 2018, 4:34 am
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i did have pms disabled, i’ve just enabled it and i get the same thing.

The rest are enabled other than…

vqplus | | Not Installed (Locally available)

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GUI Issues - Contact Manager

November 8, 2018, 4:38 am
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I’m also getting this in a red bar at the top of the dashboard.

The command "cd /var/www/html/admin/modules/pm2/node && mkdir -p /home/asterisk/.pm2 && mkdir -p /var/www/html/admin/modules/pm2/node/logs && export HOME=/home/asterisk && export PM2_HOME=/home/asterisk/.pm2 && export ASTLOGDIR=/var/log/asterisk && export ASTVARLIBDIR=/var/lib/asterisk && export PATH=$HOME/.node/bin:$PATH && export NODE_PATH=$HOME/.node/lib/node_modules:$NODE_PATH && export MANPATH=$HOME/.node/share/man:$MANPATH && /var/www/html/admin/modules/pm2/node/node_modules/pm2/bin/pm2 jlist" failed. Exit Code: 139(Segmentation violation) Working directory: /var/www/html/admin Output: ================ Error Output: ================ sh: line 1: 24807 Segmentation fault /var/www/html/admin/modules/pm2/node/node_modules/pm2/bin/pm2 jlist

File:/var/www/html/admin/libraries/Composer/vendor/symfony/process/Process.php:239

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BLF Cisco 8861 with PJSIP and others Phones

November 8, 2018, 4:56 am
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Hi,

I do not know the chan-sccp-b driver, could you explain to me ?

SourceForge tell to me: "Sorry, the permissions for this page don’t allow you to access it. "

edit: Ok, I found it: https://github.com/chan-sccp/chan-sccp/wiki/FreePBX

IF I use that driver, I have to change the SEPmac file use with the phone ?

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Dynamic On-Call Destination

November 8, 2018, 5:02 am
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Is it that you don’t understand or are not willing to understand? It was explained. It breaks features that Queues use in FreePBX. I get you’re not using those features but since Sangoma’s road map is more to the PBXact / commercial appliances. Advanced Queuing and the features that “remote login” would break are part of that.

There is also the fact that around the time that remote login feature was removed, FreePBX changed how it handles Queue Agent log ins. Coincidence? Probably not.

You also realize it’s been gone for years and there was absolutely not real blow back of it being removed. Most didn’t even know that it was gone. If this feature was so vital, it would still be around and FreePBX would have worked out it for their stuff.

The fact you don’t know that is probably part of your hang up with this issue.

100% can be done via methods provided in FreePBX to be handled in the GUI. Custom Destinations, etc.

Fact: The feature that you want is gone in new versions. It’s not coming back. You need to adapt and move on.

Fact: There is a way to make this happen. It’s an hour or two worth of work, can work for the “few” customers you have.

Fact: You have spent more time complaining about a feature that has been gone for years. You barely have a need for (a “few” clients doesn’t cut it) it. It could have been easily replicated and solved with an hour or so of work. Completely in the GUI and you would have your feature back for the new versions.

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BLF Cisco 8861 with PJSIP and others Phones

November 8, 2018, 5:08 am
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Sigh. Just keep in mind that, unless you are grossly underpaid, roughly every 3-4 hours you spend on this is equivalent to the cost of a SPA504/8G and perhaps even a side car.

Basically, the moment you hit about 6-8 hours (a days worth of work) you are costing more than the new phones.

I don’t agree with using phones that were specifically designed for a PBX system and require hacks to make work with others. Nor reverting to a channel driver that was barely supported to begin with and even less now. Sure some of those may solve your issue now but what about the future.

Replace the phones. It is the best business decision over all.

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