[Freeswitch-users] Registering with external SIP provider

[Martin Paterson]

LuKaRo,

The "sofia status profile my-sip reg" shows telephones registered to the profile, it doesn't show gateway connections.

To see whether the call arrived you could look in the FreeSWITCH log (/var/log/freeswitch/freeswitch.log), there should be entries in there whenever something happens. In addition you could run tcpcump to capture the traffic and check for the incoming INVITE.

Martin.




Martin Paterson
Development Team
Phone: 0207 953 8840
Email: martin.paterson at technologywithin.com
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-----Original Message-----
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of LuKaRo
Sent: 14 July 2020 15:31
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Registering with external SIP provider

Hi everyone,

I'm unable to register freeswitch with an external SIP provider. I gave the pages on sofia on the wiki already a good read, but I can't get it to work. One special thing is that my machine has multiple network interfaces and needs to use a specific one to connect to the external SIP provider. Therefore, I copied "external.xml" in "/opt/freeswitch/conf/sip_profiles" to "my-sip.xml" and changed the values for "rtp-ip", "sip-ip", "ext-rtp-ip" and "ext-sip-ip" to the IP address of the interface freeswitch should use, and added the "my-sip" directory to the "<X-PRE-PROCESS cmd="include" header.

Then I created "/opt/freeswitch/conf/sip_profiles/my-sip/my-sip.xml" with the following contents:

<include>
  <gateway name="my-telephony">
  <param name="proxy" value="<ip address of SIP provider>"/>
  <param name="username" value="49xxxxxxxxxx"/>
  <param name="extension" value="49xxxxxxxxxx"/>
  <param name="password" value="passwordisnotneeded"/>
  <param name="register" value="true"/>
  <param name="register-transport" value="tcp"/>
  <param name="context" value="public"/>
  </gateway>
</include>

"sofia status" on the fs_cli lists the profile as "REGED":

            my-sip    profile                 sip:mod_sofia@<local ip address>:5060    RUNNING (0)
     my-sip::my-sip    gateway                   sip:49xxxxxxxxxx@<ip address of SIP provider>    REGED

ss -an|grep <ip address of SIP provider> also lists two established TCP-connections:

tcp    ESTAB      0      0      <local ip address>:46878              <ip address of SIP provider>:5060 tcp    ESTAB      0      0      <local ip address>:5060               <ip address of SIP provider>:25976

However, "sofia status profile my-sip reg" shows 0 registrations:

Registrations:
=================================================================================================
Total items returned: 0
=================================================================================================

And when issuing a call from outside to the extension registered at the SIP provider, I just get a busy signal on the phone. Logs of the external SIP provider report "recipient unavailable".

Any ideas what could be wrong or how to further diagnose this issue? How can I check if the call reaches freeswitch?

Thanks a lot in advance,

LuKaRo

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