[Freeswitch-users] Using proxy with SIP trunk
Jose Miguel Sucasas Mejuto
jmsucasas at me.com
Sat Mar 29 22:31:20 MSK 2014
Hi all,
I’m trying to set my SIP provider on my fresh installation of FreeSWITCH. My SIP provider needs to use a proxy and authenticated username, something like this:
USERID: +3467*******
DOMAIN: ims.vodafone.es
AUTH. NAME: 214**************@ims.mnc001.mcc214.3gppnetwork.org
PROXY: 62.87.93.90:5090
I have used this configuration in directory/default/example.com.xml:
<include>
<user id="$${default_provider}">
<gateways>
<gateway name="$${default_provider}">
<!--<param name="username" value="+3467*******"/>-->
<param name="username" value="214**************@ims.mnc001.mcc214.3gppnetwork.org"/>
<param name="password" value="******"/>
<param name="from-user" value="+34672099215"/>
<param name="from-domain" value="ims.vodafone.es"/>
<param name="expire-seconds" value="600"/>
<param name="register" value="true"/>
<param name="retry-seconds" value="30"/>
<param name="extension" value="1001"/>
<param name="contact-params" value="domain_name=$${domain}"/>
<param name="context" value="public"/>
<!—Added by me -->
<param name="realm" value="ims.vodafone.es"/>
<param name="proxy" value="62.87.93.90:5090"/>
<!--<param name="register-proxy" value="62.87.93.90:5090"/>-->
</gateway>
</gateways>
<params>
<param name="password" value="******"/>
</params>
</user>
</include>
but this configuration doesn’t work: The registration is OK:
freeswitch at debrix> sofia status
Name Type Data State
=================================================================================================
192.168.1.20 alias internal ALIASED
internal profile sip:mod_sofia at 192.168.1.20:5060 RUNNING (0)
external profile sip:mod_sofia at 192.168.1.20:5080 RUNNING (0)
external::ims.vodafone.es gateway sip: 214**************@@ims.mnc001.mcc214.3gppnetwork.org at 62.87.93.90:5090 REGED
=================================================================================================
but when I try to make a call, I see the following:
freeswitch at debrix> originate sofia/gateway/ims.vodafone.es/+34677****** &playback("hello world")
2014-03-29 20:06:40.634625 [NOTICE] switch_channel.c:1050 New Channel sofia/external/+34677****** [8b2f3c51-c95a-4b12-a82a-bcbb11f8626a]
send 1046 bytes to udp/[62.87.93.90]:5090 at 19:06:40.981203:
------------------------------------------------------------------------
INVITE sip:+34677******@62.87.93.90:5090 SIP/2.0
Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH
Max-Forwards: 70
From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p
To: <sip:+34677******@62.87.93.90:5090>
Call-ID: 1a188752-3218-1232-e79c-94de8085f14a
CSeq: 57723936 INVITE
Contact: <sip:gw+ims.vodafone.es at 47.62.8.53:5080;domain_name=192.168.1.20;transport=udp;gw=ims.vodafone.es>
User-Agent: FreeSWITCH-mod_sofia/1.2.23-1~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 197
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:0000000000 at ims.vodafone.es>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1396103996 1396103997 IN IP4 47.62.8.53
s=FreeSWITCH
c=IN IP4 47.62.8.53
t=0 0
m=audio 16004 RTP/AVP 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
recv 289 bytes from udp/[62.87.93.90]:5090 at 19:06:40.985906:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 47.62.8.53:5080;received=47.62.8.53;branch=z9hG4bKQvryjNK4mDNeH;rport=5080
From: "" <sip:+34672099215 at ims.vodafone.es>;tag=pDBN9taHSHX3p
To: <sip:+34677******@62.87.93.90:5090>
Call-ID: 1a188752-3218-1232-e79c-94de8085f14a
CSeq: 57723936 INVITE
------------------------------------------------------------------------
recv 508 bytes from udp/[62.87.93.90]:5090 at 19:07:13.017222:
------------------------------------------------------------------------
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 47.62.8.53:5080;received=47.62.8.53;branch=z9hG4bKQvryjNK4mDNeH;rport=5080
From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p
To: <sip:+34677******@62.87.93.90:5090>;tag=9888
Call-ID: 1a188752-3218-1232-e79c-94de8085f14a
CSeq: 57723936 INVITE
Content-Length: 0
Contact: <sip:172.21.32.22:5070>
P-Charging-Vector: icid-value=eb15701b0bdd73150b21ba59b77106f
P-Charging-Function-Addresses: ccf="aaa://mm.ims.vodafone.es:3868;transport=tcp"
------------------------------------------------------------------------
send 334 bytes to udp/[62.87.93.90]:5090 at 19:07:13.017683:
------------------------------------------------------------------------
ACK sip:+34677******@62.87.93.90:5090 SIP/2.0
Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH
Max-Forwards: 70
From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p
To: <sip:+34677******@62.87.93.90:5090>;tag=9888
Call-ID: 1a188752-3218-1232-e79c-94de8085f14a
CSeq: 57723936 ACK
Content-Length: 0
------------------------------------------------------------------------
2014-03-29 20:07:13.014627 [NOTICE] sofia.c:6659 Hangup sofia/external/+34677****** [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
-ERR ORIGINATOR_CANCEL
2014-03-29 20:07:13.014627 [NOTICE] switch_core_session.c:1577 Session 5 (sofia/external/+34677******) Ended
2014-03-29 20:07:13.014627 [NOTICE] switch_core_session.c:1581 Close Channel sofia/external/+34677****** [CS_DESTROY]
I think that the error ( SIP/2.0 487 Request Terminated) is in the to-field of the header:
INVITE sip:+34677******@62.87.93.90:5090 SIP/2.0
Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH
Max-Forwards: 70
From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p
To: <sip:+34677******@62.87.93.90:5090>
FreeSWITCH writes the proxy instead of the realm domain (ims.vodafone.es).
Do you know how can I solve this problem?
Thanks in advance,
Regards
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