Thanks Dicko
I had several people in the office call the application and the audio was fine. I think the callers need to change their hearing aid batteries, (I can talk because I wear them myself LOL).
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Custom Recordings Playback volume level
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Kamailio and Siremis with FreePBX
Wow yeah fantastic little tool! Just installed that on both FreePBX and kamilio both, basically freePBX is transmitting but the kamilio box doesn’t get them. In fact it gets nothing, just a blank sngrep screen.
There isn’t a firewall installed on that box either…
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Kamailio and Siremis with FreePBX
On the instructions for install of dSipRouter I read this:
Install (No Proxy audio (RTP) traffic)
git clone https://github.com/dOpensource/dsiprouter.git
cd dsiprouter
./dsiprouter.sh installInstall (Proxy audio (RTP) traffic)
If you need to proxy RTP traffic then add the -rtpengine parameter. So, the command to install dSIPRouter and the RTPEngine would be
git clone https://github.com/dOpensource/dsiprouter.git
cd dsiprouter
./dsiprouter.sh install -rtpengine
I went with option 1 (No Proxy audio (RTP) traffic) - as I believe that’s only needed with NAT, is that correct?
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Kamailio and Siremis with FreePBX
hmm just tried to add the kamilio server onto a softphone (with wrong details) and sip packets appeared in sngrep which tells me FreePBX isn’t sending them properly?
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Custom Recordings Playback volume level
Sorry, I missed that, could you say it again please? 
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Kamailio and Siremis with FreePBX
A Media Proxy is indeed needed if the inbound interface is not the same as the outbound interface, this does not ONLY apply to NAT situations, but also perhaps VPN’s Internal connections etc. also. It is best to avoid it’s use if possible, you want the proxy to “hand of both” SIP and SDP both.
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Kamailio and Siremis with FreePBX
OK I won’t use that then and stick with what I have got.
Not sure why kamailio isn’t receiving what freepbx sends it though? It is receiving other test packets so I’m assuming it’s a config error on FreePBX.
For the outgoing and incoming trunk details, I have this:
type=friend
host=xx.xx.xxx.xx↧
Kamailio and Siremis with FreePBX
OK added lots more context to my trunk data, and now I’m seeing lots of traffic in my kamilio box from my FreePBX box = PROGRESS!
Now I’m getting a:
(cause 20 - Subscriber absent)
in FreePBX though…
Outgoing data:
canreinvite=yes
dtmfmode=rfc2833
host=xx.xx.xxx.xx
outboundproxy=xx.xx.xxx.xx
progressinbound=yes
qualify=300
type=peer
disallow=all
allow=ulaw
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Chan_mobile and freepbx
is it “ok” to modify conf files when using Freepbx ?
and what about the “no audio” problem when calling outside ?
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Kamailio and Siremis with FreePBX
If you change the type to “friend”, the system will use the same settings for the outbound as well as the inbound trunk. Might save you some typing…
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Whoops error on checking online for module upgrade
Run:
wget https://raw.githubusercontent.com/FreePBX/framework/release/14.0/amp_conf/htdocs/admin/libraries/modulefunctions.class.php -O /var/www/html/admin/libraries/modulefunctions.class.php && fwconsole ma downloadinstall framework --edge
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Chan_mobile and freepbx
It depends on the file. All of the “*_custom.conf” files are there for you to use and abuse. There is lots of information on this in the Wiki.
That is called “one-way audio” and even more has been written about that. I expect a Google search for “Asterisk one-way audio” will turn up several million hits, most of which will tell you about how NAT needs to be set up and how to configure your inbound router.
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Chan_mobile and freepbx
ok. after changing the context in chan_mobile.conf to “from-trunk” and creating an inbound route with ANY DID and ANY CID , now the incoming call reaches destination extention. Call gets connected but again “no audio”
it is not “one way audio”, because there’s no audio on both direction. And since this is not Sip, I think “lots of information on sip one way audio” will not help me here. This is specific to mobile channel.
is there anyone who has experience with that here ?
thanks.
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Chan_mobile and freepbx
I’ve found here something about it:
it suggests to downgrade Asterisk to v11. Is there an easy way of doing it ?
and here are similar other threads:
http://forums.asterisk.org/viewtopic.php?f=1&t=23594&start=0
Thanks.
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Yum update and distro updates
This topic was automatically closed 24 hours after the last reply. New replies are no longer allowed.
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Kamailio and Siremis with FreePBX
Im still getting
(cause 20 - Subscriber absent)
Im thinking because there is no register string?
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UCP Can't Originate Calls
Greetings to all.
When i click on a phone number to originate call this comes up.
After pressing initiate i get this:
But when i press originate, nothing happens…
Same issue over http or https, i cannot find on the wiki any settings for this that needs to be changed.
Anyone?
Thanks
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Chan_mobile and freepbx
I’m pretty sure there’s no way back to 11.
Downgrading to get this working seems like a terrible idea.
You are having one-way audio problems in both directions. I knew that when I suggested the problem was NAT and your Firewall. This is almost always problems with NAT and firewall configuration. You can fight us on it all you want, but until you are absolutely certain your firewalls, port, and NAT configurations are correct, downgrading to Asterisk 11 is a 10 year old answer.
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Kamailio and Siremis with FreePBX
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Not able to add users to groups in User Management
Any suggestions to this? Running into the same issue with UM 14.0.3.31 on 14.0.1.24 running on 12.7.4-1712-2.sng7 .
All three browsers, cleared history, etc, exhibit the same exact symptom. Click the add user to group button in group management, or vice versa in user management, gives this:
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