kein ausgehende Verbindung - Patton SN 4120 mit Freeswitch

[PaulAtreides]

Hallo,

ich kann über freeswitch mit meinem Patton 4120 keine ausgehende Verbindung aufbauen.
Im Debug Mode von Patton bekomme ich die Fehlermeldung "SIP/2.0 400 Bad Request"
Eingehende Verbindungen funktionieren. Was ist falsch?

INVITE sip:@192.168.176.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.176.6:5080;rport;branch=z9hG4bKZ4HcSyXy13a9F
Max-Forwards: 69
From: "Extension 10" <sip:[email protected]>;tag=Zr47mDvtH30pH
To: <sip:@192.168.176.4>
Call-ID: 8ad5bf27-f350-1232-818d-000024d0e264
CSeq: 68346344 INVITE
Contact: <sip:[email protected]:5080;transport=udp;gw=isdn>
User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~32bit
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: 331
X-FS-Support: update_display,send_info
Remote-Party-ID: "Extension 10" <sip:[email protected]>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1417340490 1417340491 IN IP4 192.168.176.6
s=FreeSWITCH
c=IN IP4 192.168.176.6
t=0 0
m=audio 24326 RTP/AVP 125 0 9 8 101 13
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

17:25:42  SIP_TR> [STACK] > 415 Stack: to 192.168.176.6
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.176.6:5080;rport;branch=z9hG4bKZ4HcSyXy13a9F
From: "Extension 10" <sip:[email protected]>;tag=Zr47mDvtH30pH
To: <sip:@192.168.176.4>
Call-ID: 8ad5bf27-f350-1232-818d-000024d0e264
CSeq: 68346344 INVITE
Server: Patton SN4120 2BIS4V 00A0BA0A4BD7 R6.T 2014-07-14 H323 SIP M5T SIP Stack/4.2.8.10
Content-Type: text/plain
Content-Length: 21

Invalid request line.
17:25:42  SIP_TR> [STACK] < 298 Stack: from 192.168.176.6
ACK sip:@192.168.176.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.176.6:5080;rport;branch=z9hG4bKZ4HcSyXy13a9F
Max-Forwards: 69
From: "Extension 10" <sip:[email protected]>;tag=Zr47mDvtH30pH
To: <sip:@192.168.176.4>
Call-ID: 8ad5bf27-f350-1232-818d-000024d0e264
CSeq: 68346344 ACK
Content-Length: 0

#---------------------------------------------------------------#
#
# 21.10.2014
#
#---------------------------------------------------------------#
cli version 3.20
webserver port 80 language en
sntp-client
sntp-client server primary 192.168.176.2 port 123 version 4

system
	ic voice 0
	low-bitrate-codec g711alaw64k

system
	clock-source 1 bri 0 0
	clock-source 2 bri 0 1

profile ppp default

profile tone-set default

profile voip default
	codec 1 g711alaw64k rx-length 20 tx-length 20
	codec 2 g711ulaw64k rx-length 20 tx-length 20
	fax transmission 1 relay t38-udp

profile pstn default

profile sip default
	no autonomous-transitioning

profile aaa default
	method 1 local
	method 2 none

context ip router
	interface IF_IP_WAN
	ipaddress 192.168.176.4 255.255.255.0
	tcp adjust-mss rx mtu
	tcp adjust-mss tx mtu

context cs switch
	national-prefix 0
	international-prefix 00

	routing-table called-e164 RT_ISDN_TO_SIP
	route T dest-interface IF_SIP

	routing-table calling-e164 RT_SIP_TO_ISDN

interface isdn IF_ISDN_00
	route call dest-table RT_ISDN_TO_SIP
	isdn-date-time

interface isdn IF_ISDN_01
	route call dest-table RT_ISDN_TO_SIP
	isdn-date-time

interface sip IF_SIP
	bind context sip-gateway GW_SIP
	route call dest-table RT_SIP_TO_ISDN
	remote 192.168.176.6 

service hunt-group SV_HUNT_PSTN
	drop-cause normal-unspecified
	drop-cause no-circuit-channel-available
	drop-cause network-out-of-order
	drop-cause temporary failure
	drop-cause switching-equipment-congestion
	drop-cause access-info-discarded
	drop-cause circuit-channel-not-available
	drop-cause resources-unavailable
	route call 1 dest-interface IF_ISDN_00
	route call 2 dest-interface IF_ISDN_01

context cs switch
	no shutdown

authentication-service AUTH_SVC
	username smartnode password ********

location-service LOCATION_SVC
	domain 1 192.168.176.6 5080
	identity smartnode
		authentication outbound
      			authenticate 1 authentication-service AUTH_SVC username smartnode
    		registration outbound
	    	registrar 192.168.176.6 5080
    		lifetime 3600
    		register auto

context sip-gateway GW_SIP
	no shutdown

	interface IF_WAN
	bind interface IF_IP_WAN context router port 5060

context sip-gateway GW_SIP
	no shutdown

port ethernet 0 0
	medium auto
	encapsulation ip
	bind interface IF_IP_WAN router
	no shutdown

port bri 0 0
	clock auto
	encapsulation q921
	
	q921
		protocol pp
		uni-side auto
		encapsulation q931
	q931
		protocol dss1
		uni-side user
		bchan-number-order ascending
		encapsulation cc-isdn
		bind interface IF_ISDN_00 switch
port bri 0 0
	no shutdown

port bri 0 1
	clock auto
	encapsulation q921
	
	q921
		protocol pp
		uni-side auto
		encapsulation q931
	q931
		protocol dss1
		uni-side user
		bchan-number-order ascending
		encapsulation cc-isdn
		bind interface IF_ISDN_01 switch
port bri 0 1
	no shutdown

SIP PROFILE

<include>
 <gateway name="isdn">
  <param name="proxy" value="192.168.176.4"/>
  <param name="register" value="false"/>
  <param name="caller-id-in-from" value="true"/>
  <param name="username" value="smartnode"/>
  <param name="password" value="********"/>
 </gateway>
</include>

<include>
 <extension name="isdn-outbound">
  <condition field="destination_number" expression="^0(.*)$">
   <action application="bridge" data="sofia/gateway/isdn/$1"/>
  </condition>
 </extension>
</include>

 

[sunseb]

Hallo,

Ich kenne mich mit freeswitch leider nicht aus.

Im eingehenden INVITE fehlt aber die Zielrufnummer:

INVITE sip:???@192.168.176.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.176.6:5080;rport;branch=z9hG4bKZ4HcSyXy13a9F
Max-Forwards: 69
From: "Extension 10" <sip:[email protected]>;tag=Zr47mDvtH30pH
To: <sip:???@192.168.176.4>

Folglich kann die Patton auch kein Gespräch aufbauen...

Ein weiterer Punkt wäre das Interface sip IF_SIP welches die calls zu RT_SIP_TO_ISDN leitet. Diese Tabelle ist aber leer...

Gruß, S