Hello, thanks for your comment. Perhaps I was a little unspecific. So here are some more details: I'm a beginner concerning VOIP (I reached a state where I am able to connect VOIP endpoints, develop a dialplan and connect to an external provider) but a very experienced network and Linux user. Because I would like to learn Freeswitch from scratch I started with the minimal configuration of the Jessie distribution. I added a gateway definition to the external profile (tried the internal profile, too - same behaviour): <include> <gateway name="x"> <param name="realm" value="IP ADDRESS"/> <param name="username" value="USERID" /> <param name="password" value="PASSWORD" /> <param name="register" value="true" /> </gateway> </include> When I start Freeswitch a connaction attempt is made. A REGISTER is send to the HIPATH, a TRYING and then I receive an answer 404 NOTFOUND. If I use X-lite and connect to the HIPATH the connection attempt is successful. So the SIP account of the HIPATH seems to work. My sofia.conf.xml is: <configuration name="sofia.conf" description="sofia endpoint"> <global_settings> <param name="log-level" value="0"/> <param name="tracelevel" value="DEBUG"/> </global_settings> <profiles> <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/> </profiles> </configuration> And my external.xml is: <profile name="external"> <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <settings> <param name="sip-port" value="$${external_sip_port}"/> <param name="auth-calls" value="false"/> <param name="debug" value="0"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> </settings> </profile> The console says: Failed registration. Any hints on what is missing? I looked up the vanilla configuration but there seems to be no promising additional options. Thanks. Dieter Am 28.09.2016 um 17:24 schrieb Peter Steinbach:
Hello Dieter,
sure this should work. Registering a Freeswitch as a SIP client to an external SIP server can be done through a gateway definition, as you pointed out. The first step is to see, if the gateway is registered to the Hipath ("sofia status" should show REGED for the gateway). If yes, you're almost done. (I assume, you have to register to the Hipath, right?) I think, that Freeswitch does not offer Subscribes in the external profile, only in the default internal profile. Do you receive an INVITE from the Hipath? Also note, that Freeswitch's external profile is on Port 5080 instead of 5060, so Hipath has to send to 5080. On Linux, you may use ngrep to grep the SIP traffic: "ngrep -d any port 5080 -W byline").
Best regards Peter
On 09/28/16 14:22, Scholz, Dieter wrote:
Hello,
this is my problem:
I would like to use Freeswitch as a gateway between our Hipath PBX and an external SIP provider.
For that I created a Hipath SIP account that is working (tested with X-lite). Now I would like to let Freeswitch act as a client using this SIP account. Whenever there is an incoming call Freeswitch should act on it and forward this call to an external SIP provider.
Is this possible? Does that make sense? Are there alternatives?
At the moment I'm stuck to configure Freeswitch as a client. I added the Hipath SIP account as a gateway. When I compare the SIP traffic of X-lite and Freeswitch using Wireshark I found out that Freeswitch dows not offer the 'Subscribe' option. Is this the reason why it fails?
Can you help me?
Thanks in advance.
Dieter
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