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Kamailio and Siremis with FreePBX

January 17, 2018, 9:08 am
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Only dicko thinks it’s so easy, and don’t be fooled - he has probably been working with Kamailio / openser for quite a long time. :slight_smile: He just wants you to know how clever he is…

My own opinion is that the scripting language of Kamailio or its cousin OpenSIPS is tricky and requires more than a basic understanding of SIP, certainly more understanding than you would need to successfully stand up a FreePBX server, as you are holding the SIP building blocks in your hands and have to know how to put them together to make something useful happen.

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No Incoming calls

January 17, 2018, 9:11 am
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I’ve never used one of these, but the PSTN-To-VOIP Gateway settings don’t look right.

In a general sense, inbound and outbound calling are actually different processes, so getting outgoing calls to work is largely unrelated to inbound calling working. If you are not getting notified of an inbound call, it’s either because the SPA3102 isn’t set up to connect to your PBX, or your PBX is not set up to answer the incoming SIP connection from the SPA3102. With your setup (as documented), I’m pretty sure you have your SIP information set incorrectly on the SPA, and I’m not confident your trunk for the inbound calls is set up correctly as well.

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Can't make outbound calls via sip trunk

January 17, 2018, 9:13 am
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Sanitize and post your trunk setup.

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Kamailio and Siremis with FreePBX

January 17, 2018, 9:14 am
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I’m hitting the ground running with dSIPRouter. How do you setup an IP-only trunk on FreePBX?

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TLS Cerificate

January 17, 2018, 9:15 am
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It isn’t that you’re wrong, but that’s not how the system works. The certs on the server provide all of the security you will need, especially if you instantiate a VPN between the phone and the server.

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Error[6641] chan_sip.c:4273 Serious Network Trouble; __sip_xmit returns error for pkt data

January 17, 2018, 9:19 am
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If I understand the problem, setting the STUN server is the right solution for this, since the packets going back to the client have the wrong address otherwise. Note that you should be able to set the “external” address on the Zulu side manually if the “external” IP of the network doesn’t change, but setting the STUN server will work to set that external address in either case.

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New VM recordings

January 17, 2018, 9:24 am
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She’s going to be very angry that you called her a TTS Engine. All of the system prompts are recorded by a real person (whose name has escaped me). You can hire her to record additional prompts if you want the system to be self-integrated.

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Kamailio and Siremis with FreePBX

January 17, 2018, 9:26 am
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Pretty well all you need outgoing is

type=friend
host=192.168.1.1

Incoming just needs similar, or disable just anonymous calls to test and set an inbound route

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Chan_mobile and freepbx

January 17, 2018, 9:30 am
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This usually indicates you are trying to pick up an extension from Chan-SIP on a PJ-SIP port (or vice versa).

The error message says that you tried to send the call to a context called “from-MySipphone” with a ‘s’ matching extension and priority of ‘1’, and that combination of context, extension, and priorotity doesn’t actually exist. Look in your extensions*conf files and see if you can find the ‘from-MySipphone’ context. If not, you may need to add it to your extensions_custom.conf file.

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Kamailio and Siremis with FreePBX

January 17, 2018, 9:30 am
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That’s where Siremis comes inm, it taks all that out of the equation, have you tried it yet?

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Multiple calls not following dialplans

January 17, 2018, 9:35 am
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My next guess would be that the cell# is capturing the call and it isn’t returning to the PBX. I had a problem like this with a cell phone and it was because the cell was “expiring” the call and routing it to VM before the system would have expired the call.

While sending a call to a cell phone sometimes works, it’s not always going to. If the cell, for example, marks the call as “complete” with the PBX, even if you didn’t answer it, the call is considered “answered” and you will never return to the error-logic leg.

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Kamailio and Siremis with FreePBX

January 17, 2018, 9:37 am
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Yes I tried siremis, but couldn’t figure it out - incredibly over complicated unfortunately.

I’ve setup dSIProuter and it appears to be working OK, however now when I configure FreePBX trunk like you show above, when dialling out it tries to make the call but comes back saying
"The person isn’t answering" and in the logs it’s showing Retransmission Error and never responds.

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Remove local network from Asterisk SIP settings?

January 17, 2018, 9:38 am
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server running:
freepbx 14.0.1.24
asterisk 13.18.5

I cannot remove a local network from the asterisk sip settings page. GUI says blank fields will be ignored, however hitting submit while blank fields are present returns error and cannot save changes.

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Kamailio and Siremis with FreePBX

January 17, 2018, 9:40 am
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No, but thanks for the lead. I’ll give it a look sometime.

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Kamailio and Siremis with FreePBX

January 17, 2018, 9:45 am
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My FreePBX never hits the Kamilio server (or at least the Kamilio server never responds)

I will try and disable FreePBX firewall to test.

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Whoops error on checking online for module upgrade

January 17, 2018, 9:52 am
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At some point after updates yesterday (new rev of php56 module via yum/system updates and a few new module updates via the traditional module updates, including a freepbx framework update), my web interface is returning a whoops error when checking online for module updates. When using the command line (fwconsole ma update all), I get the following:

PHP Fatal error: Class ‘Rhumsaa\Uuid\Uuid’ not found in /var/www/html/admin/libraries/modulefunctions.class.php on line 2661
Whoops\Exception\ErrorException: Class ‘Rhumsaa\Uuid\Uuid’ not found in file /var/www/html/admin/libraries/modulefunctions.class.php on line 2661
Stack trace:

  1. Whoops\Exception\ErrorException->() /var/www/html/admin/libraries/modulefunctions.class.php:2661

I am also getting incorrect info on the Asterisk info page (shows no trunks/extensions connected, etc) - but actual phone system functionality is intact (can make/receive internal calls, external calls, vm access, all OK.) UCP also seems OK.

I am running distro version 12.7.4-1712-2.sng7, freepbx version 14.0.1.28, asterisk version 15.1.5

Any thoughts on how to resolve? System was behaving properly before updates yesterday …

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Whoops error on checking online for module upgrade

January 17, 2018, 9:54 am
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Oh, I also have the following error on my dashboard:

Cronmanager encountered 1 Errors

The following commands failed with the listed error
/var/lib/asterisk/bin/module_admin listonline > /dev/null 2>&1 (255)

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Kamailio and Siremis with FreePBX

January 17, 2018, 9:59 am
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sngrep is a very good tool for a quick look, run it on both boxes and watch them

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New VM recordings

January 17, 2018, 10:18 am
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Remove local network from Asterisk SIP settings?

January 17, 2018, 10:29 am
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Upgrade the SIP settings module to edge with:

fwconsole ma --edge upgrade sipsettings
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