Hello!
I'm trying to create a Multifon trunk for Asterisk via FreePBX. However, whatever settings you use, in the end, the report (sip show registry (in the console) or Reports-> Asterisk Info-> Chan_SIP Info (in FreePBX)) goes "state: Request Sent". The basic settings were:

Peers Details :
username = 792 <number>
type = peer
secret = <password>
qualify = yes
notransfer = no
insecure = port invite
host = 193.201.229.35
fromuser = 792 <number>
fromdomain = multifon.ru
disallow = all
allow = ulaw
dtmfmode = inband
User Details :
type = user
context = from-pstn
dtmfmode = inband
disallow = all
allow = ulaw
Register String :
792 <number>: <password>: 792 <number> @ 193.201.229.35 / 792 <number>

They were taken from http://trustore.ru/article/asterisk/213-multifon-in-freepbx
From the softphone, calls through the Multiphone are available, on the website https://sm.megafon.ru/sm/client/routing set status 2 (for receiving calls through the softphone).
Please help me, what is wrong?

  • one
    Good day! In the Register String: 792<номер>:пароль>:792<пароль>@193.201.229.35/792<номер> line Register String: 792<номер>:пароль>:792<пароль>@193.201.229.35/792<номер> you probably have a typo. It should be something like this: 792<номер>:<пароль>:792<номер>@193.201.229.35/792<номер> Otherwise, check the NAT settings, sip set debug on in the asterisk console and see what happens there. - StuxForce
  • you are right, wrong) - nita
  • in console after sip set debug on was output by Really destroing sip dialog Method: OPTIONS - nita
  • one
    state: Request Sent means that your server sent a request but has not received a response yet. Do you have a server for NAT? If so, is the externip parameters set correctly in sip.conf? In general, in the debug, you need to catch the REGISTER message from your server to the multifon server. There everything will be immediately visible. - StuxForce
  • one
    I repeat, the problem is most likely with the configuration of NAT. You need to first configure the externip , localnet and nat parameters in sip.conf. Your server most likely sends to the operator in the REGISTER message a local address (internal) and should substitute an external one. Therefore, the operator can not send you a response packet. Fully quote the message REGISTER, there are lines To: From: Contact :, what's in them? - StuxForce

2 answers 2

You need to properly configure asterisk to work with NAT. You can read a little about it, for example, here: https://voipnotes.ru/nastroika-asterisk-i-nat/

What file will need to be edited? the one that is in / etc / asterisk

The main config. the pure asterisk file is in /etc/asterisk/sip.conf In FreePBX, unfortunately, I will not tell you. Perhaps there is a setting in graphical mode. http://wiki.freepbx.org/display/FPG/Asterisk+SIP+Settings+User+Guide You will also need to configure port forwarding for RTP on your asterisk server on the router

    Try this:

    register => 792xxxxxxxx@multifon.ru: pasSWORD123: 792xxxxxxxx@sbc.megafon.ru: 5060 / 792xxxxxxx

    taken from the manual megaphone