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
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@technologywithin.com Chevron Business Park, Limekiln Lane, Southampton, Hampshire, SO45 2QL Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K Registered in England No: 5964349 | VAT Number: GB 902 5369 37 -----Original Message----- From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> On Behalf Of LuKaRo Sent: 14 July 2020 15:31 To: freeswitch-users@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
On 7/14/20 4:31 PM, LuKaRo wrote:
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. [...]
Hi everyone, it looks like my mail took over a week for processing. I just got the confirmation that it has been posted to the list, after sending it last week. However, my issue is already solved, so you can consider this thread closed :) I don't want to waste anyones time on an already solved issue :) Thanks (and sorry for the noise), LuKaRo
participants (2)
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LuKaRo
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Martin Paterson