Making outbound calls with UserAgentGui and UserAgentCli

[kwaggaz]

Hi,

I have successfully registered both the UserAgentGui and UserAgentCli to an Asterisk server. I can call both the UserAgentGui and UserAgentCli phones using another SIP phone, and I am able to answer the inbound calls and have successful conversations. However, I am unable to make outbound calls to another SIP phone on the same Asterisk server. I receive the following error:

Exception in thread "AWT-EventQueue-0" java.lang.NullPointerException: Cannot invoke "org.mjsip.sdp.SdpMessage.toString()" because "session_descriptor" is null
	at org.mjsip.sip.dialog.InviteDialog.invite(InviteDialog.java:315)
	at org.mjsip.sip.dialog.InviteDialog.inviteWithoutOffer(InviteDialog.java:367)
	at org.mjsip.sip.call.ExtendedCall.call(ExtendedCall.java:135)
	at org.mjsip.sip.call.Call.call(Call.java:183)
	at org.mjsip.sip.call.Call.call(Call.java:176)
	at org.mjsip.ua.UserAgent.call(UserAgent.java:180)
	at org.mjsip.ua.UserAgent.call(UserAgent.java:170)
	at org.mjsip.ua.UserAgent.call(UserAgent.java:156)
	at org.mjsip.phone.gui.UserAgentGui.jButton1_actionPerformed(UserAgentGui.java:431)
	at org.mjsip.phone.gui.UserAgentGui$2.actionPerformed(UserAgentGui.java:310)

I run both programs with the following parameters: -f ../mjsip.cfg --prefer-ipv4 true --address sip:[email protected]:5060
and with the following config file:

# 
#                   MjSip-1.7 Configuration File
# ________________________________________________________________
#
# This file inclues all MjSip configuration parameters.
# MjSip parameters are organized into 6 sections:
#  o Section 1: SipStack base configuration
#  o Section 2: Logs
#  o Section 3: SipProvider configuration
#  o Section 4: Server configuration
#  o Section 5: UA configuration
#  o Section 6: SBC configuration
#



# ________________________________________________________________
#
# Section 1: SipStack base configuration
# ________________________________________________________________
#
# Normally, you do not have to change the base configuration,
# and you can go directly to Section 2.
# SIP and transport layer configurations are handled in Section 3. 
#

# Default SIP port.
# Note that this is not the port used by the running stack, but simply the standard default SIP port.
# Normally it sould be set to 5060 as defined by RFC 3261. Using a different value may cause
# some problems when interacting with other unaware SIP UAs. 
# Default value: default_port=5060
#default_port=5060

# Default SIP port for TLS transport (SIPS).
# Note that this is not the port used by the running stack, but simply the standard default SIPS port.
# Normally it sould be set to 5061 as defined by RFC 3261. Using a different value may cause
# some problems when interacting with other unaware SIP UAs.
#default_tls_port=5061

# Default supported transport protocols.
# Default value: default_transport_protocols=udp,tcp
#default_transport_protocols=udp

# Default max number of contemporary open transport connections.
# Default value: default_nmax_connections=32
#default_nmax_connections=0

# Whether adding 'rport' parameter on via header fields of outgoing requests.
# Default value: use_rport=yes
#use_rport=no

# Whether adding (forcing) 'rport' parameter on via header field of incoming responses.
# Default value: force_rport=no
#force_rport=yes

# Starting retransmission timeout (milliseconds); called T1 in RFC2361; they suggest T1=500ms 
# Default value: retransmission_timeout=500
#retransmission_timeout=2000 

# Maximum retransmission timeout (milliseconds); called T2 in RFC2361; they suggest T2=4sec 
# Default value: max_retransmission_timeout=4000  
#max_retransmission_timeout=4000  

# Transaction timeout (milliseconds); RFC2361 suggests 64*T1=32000ms 
# Default value: transaction_timeout=32000
#transaction_timeout=10000    

# Clearing timeout (milliseconds); T4 in RFC2361; they suggest T4=5sec 
# Default value: clearing_timeout=5000
#clearing_timeout=5000

# Default max-forwards value (RFC3261 recommends value 70).
# Default value: max_forwards=70
#max_forwards=10

# Whether using only one thread for all timer instances (less precise but more efficient).
# Default value: single_timer=yes
#single_timer=no

# Whether at UAS side automatically sending (by default) a 100 Trying on INVITE.
# Default value: auto_trying=yes
#auto_trying=no

# Whether 1xx responses create an "early dialog" for methods that create dialog.
# Default value: early_dialog=no
#early_dialog=yes

# Default 'expires' time in seconds. RFC2361 gives as default value expires=3600 
# Default value: default_expires=3600
#default_expires=1800

# UA info included in request messages (in the User-Agent header field).
# Use 'NONE' string or let it blank if the User-Agent header filed must be added.
# Default value: ua_info=<the mjsip release>
# ua_info=NONE

# Server info included in request messages (in the Server header field).
# Use 'NONE' string or let it blank if the Server header filed must be added.
# Default value: server_info=<the mjsip release>
# server_info=NONE

# Starting registration timeout (msecs) after a registration failure due to request timeout.
# Default value: regc_min_attempt_timeout=60*1000 (that is 1min)
#regc_min_attempt_timeout=30*1000

# Maximum registration timeout (msecs) after a registration failure due to request timeout.
# Default value: regc_max_attempt_timeout=900*1000 (that is 15min)
#regc_max_attempt_timeout=600*1000

# Maximum number of consecutive registration authentication attempts before giving up.
# Default value: regc_auth_attempts=3
#regc_auth_attempts=5

# Whether forcing this node to stay within the dialog route as peer,
# by means of the insertion of a RecordRoute header.
# This is a non-standard behaviour and is normally not necessary.
# Default value: on_dialog_route=no
#on_dialog_route=yes



# ________________________________________________________________
#
# Section 3: SipProvider configuration
# ________________________________________________________________
#
# Change these parameters in order to customize the SIP transport layer.
# Usually you have to deal with some of these configuration parameters.
#

# Via IP address or fully-qualified domanin name (FQDN).
# Use 'AUTO-CONFIGURATION' for auto detection (or let it undefined).
# Default value: via_addr=AUTO-CONFIGURATION
via_addr=10.252.1.213

# Local SIP port.
# Default value: host_port=5060
#host_port=5060

# Network interface (IP address) used by SIP.
# Use 'all-interfaces' for binding SIP to all interfaces (or let it undefined).
# Default value: host_ifaddr=ALL-INTERFACES
#host_ifaddr=ALL-INTERFACES

# List of enabled transport protocols (the first protocol is used as default)
# Default value: transport_protocols=udp
transport_protocols=udp

# List of transport ports, ordered as the corresponding transport_protocols.
#transport_ports=5060,5060

# Max number of contemporary open transport connections.
# Default value: nmax_connections=32
#nmax_connections=0

# Outbound proxy URL ([sip:]host_addr[:host_port][;transport=proto]).
# Use 'NONE' for not using an outbound proxy (or let it undefined).
# Default value: outbound_proxy=NONE
#outbound_proxy=proxy.example.net:5060

# Whether logging all packets (including non-SIP keepalive tokens).
# Default value: log_all_packets=no
#log_all_packets=yes

# For TLS. Whether all client and server certificates should be considered trusted.
# By default, trust_all=no
#trust_all=yes

# For TLS. Path of the folder where trusted certificates are placed.
# All certificates (with file extension ".crt") found in this folder are considered trusted.
# By default, the folder "./cert" is used.
#trust_folder=cert

# For TLS. Absolute file name of the certificate (containing the public key) of the local node.
# The file name includes the full path starting from the current working folder.
# By default, the file "./cert/ssl.crt" is used.
#cert_file=cert/tls.crt

# For TLS. Absolute file name of the private key of the local node.
# The file name includes the full path starting from the current working folder.
# By default, the file "./cert/ssl.key" is used.
#key_file=cert/tls.key


# ________________________________________________________________
#
# Section 4: Server configuration
# ________________________________________________________________
#
# Change these parameters in order to customize the Server behaviour.
# You need to edit this section only if you are using a MjSip Server.
#

# The domain name(s) that the server administers.
# It lists the domain names for which the Location Service wmaintains user bindings. 
# Use 'auto-configuration' for automatic configuration of the domain name.
#domain_names=example.net example.com
#domain_names=AUTO-CONFIGURATION

# Whether consider any port as valid local domain port (regardless which sip port is used).
# Default value: domain_port_any=no
#domain_port_any=yes

# Whether the Server should act as Registrar (i.e. respond to REGISTER requests).    
# Default value: is_registrar=yes
#is_registrar=no

# Maximum expires time (in seconds).
# Default value: expires=3600
#expires=1800

# Whether the Registrar can register new users (i.e. REGISTER requests from unregistered users).    
# Default value: register_new_users=yes
#register_new_users=no

# Whether the Server relays requests for (or to) non-local users.   
# Default value: is_open_proxy=yes
#is_open_proxy=no

# The type of location service.
# You can specify the location service type (e.g. local, ldap, radius, mysql)
# or the class name (e.g. local.server.LocationServiceImpl).
# Default value: location_service=local
#location_service=ldap

# The name of the location DB.
# Default value: location_db=users.db
#location_db=config/users.db

# Whether location DB has to be cleaned at startup.
# Default value: clean_location_db=no
#clean_location_db=yes

# Whether the Server authenticates local users.   
# Default value: do_authentication=no
#do_authentication=yes

# Whether the Proxy authenticates users.
# Default value: do_proxy_authentication=no
#do_proxy_authentication=yes

# The authentication scheme.
# You can specify the authentication scheme name (e.g. Digest, AKA, etc.)
# or the class name (e.g. local.server.AuthenticationServerImpl).
# Default value: authentication_scheme=Digest
#authentication_scheme=AKA

# The authentication realm.
# If not defined or equal to 'NONE' (default), the used via address is used instead.
# Default value: authentication_realm=NONE
#authentication_realm=example.net

# The type of authentication service.
# You can specify the authentication service type (e.g. local, ldap, radius, mysql)
# or the class name (e.g. local.server.AuthenticationServiceImpl).
# Default value: authentication_service=local
#authentication_service=ldap

# The name of the authentication DB.
# Default value: authentication_db=aaa.db
#authentication_db=config/aaa.db

# Whether the server should stay in the signaling path (uses Record-Route/Route).
# Default value: on_route=no
#on_route=yes

# Whether implementing the RFC3261 Loose Route (or RFC2543 Strict Route) rule.
# Default value: loose_route=yes
#loose_route=no

# Whether checking for loops before forwarding a request (Loop Detection). In RFC3261 it is optional.
# Default value: loop_detection=yes
#loop_detection=no

# Proxy transaction timeout (in milliseconds), that corresponds to Timer "C" of RFC2361;
# RFC2361 suggests C>3min=180000ms.
# Default value: proxy_transaction_timeout=180000
#proxy_transaction_timeout=120000

# Sequence of pairs of username or phone prefix and corresponding nexthop address.
# It provides static rules for proxying number-based SIP-URL the server is responsible for.
# Use "default" (or "*") as default prefix.
# Example:
#   server is responsible for the domain 'example.com'
#   phone_proxying_rules={prefix=0123,nexthop=127.0.0.2:7002} {prefix=4567,nexthop=127.0.0.2:7003} {prefix=*,nexthop=127.0.0.3:7004}
#   a message with recipient 'sip:[email protected]' is forwarded to 'sip:[email protected]:7002'
#authenticated_phone_proxying_rules={prefix=0123,nexthop=127.0.0.2:7002}
#phone_proxying_rules={prefix=4567,nexthop=127.0.0.2:7003} {prefix=*,nexthop=127.0.0.3:7004}

# Sequence of pairs of destination domain and corresponding nexthop address.
# It provides static rules for proxying domain-based SIP-URL the server is NOT responsible for.
# It make the server acting (also) as 'Interrogating' Proxy, i.e. I-CSCF in the 3G networks.
# Example: <br>
#   server is responsible for the domain 'example.com'
#   domain_proxying_rules={domain=domain1.foo,nexthop=proxy.example.net:5060}
#   a message with recipient 'sip:[email protected]' is forwarded to 'sip:[email protected]:5060'
#authenticated_domain_proxying_rules={domain=domain1.foo,nexthop=192.168.0.4:5060}
#domain_proxying_rules={domain=domain2.foo,nexthop=192.168.0.5:5060}



# ________________________________________________________________
#
# Section 5: UA configuration
# ________________________________________________________________
#
# Change these parameters in order to customize the UA profile.
# You need to edit this section only if you are using a MjSip UA or
# you are managing 3PCC services.
#

# Display name for the user.
# It is used in the user's AOR registered to the registrar server and used as From URL.
display_name=777

# User's name.
# It is used to build the user's AOR registered to the registrar server and used as From URL.
user=777

# Fully qualified domain name (or address) of the proxy server.
# It is part of the user's AOR registered to the registrar server and used as From URL.
# If 'proxy' is not defined, the 'registrar' value is used in its place.
# If 'registrar' is not defined, the 'proxy' value is used in its place.
#proxy=proxy.example.net

# Fully qualified domain name (or address) of the registrar server.
# It is used as recipient for REGISTER requests.
# If 'registrar' is not defined, the 'proxy' value is used in its place.
# If 'proxy' is not defined, the 'registrar' value is used in its place.
registrar=10.252.1.213

# Contact URL.
# If not defined (default), it is formed as sip:user@via_addr:host_port.
# Normally you can leave it undefined.
#contact_url=sip:[email protected]:5070
#contact_url=sip:[email protected]:5060

# User's name used for server authentication.
auth_user=777

# User's realm used for server authentication.
auth_realm=asterisk

# User's passwd used for server authentication.
auth_passwd=password

# Absolute path (or complete URL) of the jar archive, where various UA media (gif, wav, etc.) are stored.
# Use value 'NONE' for getting resources from external folders.
# Default value: ua_jar=lib/ua.jar
#ua_jar=lib/ua.jar

# Absolute path (or complete URI) of the folder containing UA's resources.
# By default, the local folder "resources" is used.
#res_path=resources

# Relative path of UA media resources (gif, wav, etc.) within the UA jar file or within the resources folder. 
# By default, the folder "media/local/ua" is used.
#media_path=media/local/ua

# Absolute path (or complete URL) of the buddy list file where the buddy list is and loaded from (and saved to).
# Default value: buddy_list_file=buddy.lst
#buddy_list_file=config/buddy.lst

# Whether registering with the registrar server.
# Default value: do_register=no
do_register=yes

# Whether unregistering the contact address.
# Default value: do_unregister=no
#do_unregister=yes

# Whether unregistering all contacts beafore registering the contact address.
# Default value: do_unregister_all=no
#do_unregister_all=yes

# Expires time.
# Default value: expires=3600
# expires=1800

# Rate of keep-alive tokens (datagrams) sent toward the outbound proxy (if present) or toward the registrar server.
# Its value specifies the delta-time (in millesconds) between two keep-alive tokens.
# Set keepalive_time=0 for not sending keep-alive datagrams.
# Default value: keepalive_time=0
# keepalive_time=8000

# Automatic call a remote user secified by the 'call_to' value.
# Use value 'NONE' for manual calls (or let it undefined).
# Default value: call_to=NONE
#call_to=sip:[email protected]   
  
# Response time in seconds; it is the maximum time the user can wait before responding to an incoming call;
# after such time the call is automatically declined (refused).
# Default value: refuse_time=20

# Automatic answer time in seconds; time<0 corresponds to manual answer mode.
# Default value: accept_time=-1       
#accept_time=0

# Automatic hangup time (call duartion) in seconds; time<=0 corresponds to manual hangup mode.
# Default value: hangup_time=-1
#hangup_time=10

# Automatic call transfer time in seconds; time<0 corresponds to no auto transfer mode.
# Default value: transfer_time=-1
#transfer_time=10

# Automatic re-inviting time in seconds; time<0 corresponds to no auto re-invite mode.
# Default value: re_invite_time=-1
#re_invite_time=10

# Redirect incoming call to the secified url.
# Use value 'NONE' for not redirecting incoming calls (or let it undefined).
# Default value: redirect_to=NONE
#redirect_to=sip:[email protected]

# Transfer calls to the secified url.
# Use value 'NONE' for not transferring calls (or let it undefined).
# Default value: transfer_to=NONE
#transfer_to=sip:[email protected]

# No offer in the invite.
# Default value: no_offer=no
no_offer=yes

# Whether using audio.
# Default value: audio=no
audio=yes

# Whether using video.
# Default value: video=no
#video=yes

# Whether looping the received media streams back to the sender.
# Default value: loopback=no
#loopback=yes

# Whether playing in receive only mode.
# Default value: recv_only=no
#recv_only=yes

# Whether playing in send only mode.
# Default value: send_only=no
#send_only=yes

# Whether playing a test tone in send only mode.
# Default value: send_tone=no
#send_tone=yes

# Audio file to be played.
# Default value: send_file=NONE
#send_file=audio1.in

# Audio file to be recored.
# Default value: recv_file=NONE
#recv_file=audio1.out

# Media address (use it ONLY if you want to specify a media address different from the via address).
#media_addr=10.252.1.213

# First media port (use it ONLY if you want to specify media ports different from those specified by the 'media' parameter).
#media_port=7000

# Whether using symmetric_rtp.
# Default value: symmetric_rtp=no
#symmetric_rtp=yes

# Media descriptors:
# One or more 'media' (or 'media_desc') parameters specify for each supported media: the media type, port, and protocol/codec.
# Zero or more 'media_spec' parameters can be used to specify media attributes such as: codec name, sample rate, and frame size.
# Examples:
#   media=audio 4000 rtp/avp
#   media_spec=audio 0 PCMU 8000 160
#   media_spec=audio 8 PCMA 8000 160
#   media_spec=audio 101 G726-32 8000 80
#   media_spec=audio 102 G726-24 8000 60
#   media=video 3002 rtp/avp
#   media_spec=video 101
# Alternatively media attributes can be specified also within the 'media' parameter as comma-separated list between brackets.
# Examples:
#   media=audio 4000 rtp/avp {audio 0 PCMU 8000 160, audio 8 PCMA 8000 160}
#   media=video 3002 rtp/avp {video 101}
# Default values:
#media=audio 4000 rtp/avp
#media_spec=audio 0 PCMU 8000 160
#media=video 4002 rtp/avp
#media_spec=video 101
#media=audio 12506 RTP/AVP 0 8 101
media=audio 12858 RTP/AVP 0 101 {0 PCMU 8000 160, 8 PCMA 8000 160}

# Whether using JMF for audio streaming.
# Default value: use_jmf_audio=no
#use_jmf_audio=yes

# Whether using JMF for video streaming.
# Default value: use_jmf_video=yes
#use_jmf_video=no

# Whether using RAT (Robust Audio Tool) as audio sender/receiver.
# Default value: use_rat=no
#use_rat=yes

# RAT command-line executable.
# Default value: bin_rat=rat
#bin_rat=c:\programmi\mbone\rat

# Whether using VIC (Video Conferencing Tool) as video sender/receiver.
# Default value: use_vic=no
#use_vic=yes

# VIC command-line executable.
# Default value: bin_vic=vic
#bin_vic=c:\programmi\mbone\rat


# ________________________________________________________________
#
# Section 6: SBC
# ________________________________________________________________
#
# Change these parameters in order to customize the SessionBorderController (SBC) behaviour.
# You need to edit this section only if you are using a SBC.
#

# Maximum time that the UDP relay remains active without receiving UDP datagrams (in milliseconds).
# Default value: relay_timeout=60000

# Refresh time of address-binding cache (in milliseconds)
# Default value: binding_timeout=3600000

# Minimum time between two changes of peer address (in milliseconds)
# Default value: handover_time=0

# Rate of keep-alive datagrams sent toward all registered UAs (in milliseconds).
# Set keepalive_time=0 to disable the sending of keep-alive datagrams.
# Default value: keepalive_time=0

# Whether sending keepalive datagram to all contacted UAs (also toward non-registered UAs)
# Default value: keepalive_aggressive=no

# Minimum inter-packet departure time (in milliseconds).
# Set interpacket_time=0 to disable this option.
# Default value: interpacket_time=0 

# Whether intercepting media traffics.
# Default value: do_interception=no

# Whether injecting new media flows.
# Default value: do_active_interception=no

# Sink address for media traffic interception.
# Default value: sink_addr=127.0.0.1

# Sink port for media traffic interception.
# Default value: sink_port=0

# Media address.
# Default value: media_addr=0.0.0.0

# Available media ports.
# Default value: media_ports=41000-41499

# Backend proxy where all requests not coming from it are passed to. 
# It can be specified as FQDN or host_addr[:host_port].
# Use 'NONE' for not using a backend proxy (or let it undefined).
# Default value: backend_proxy=NONE
#backend_proxy=127.0.0.2:5069

Your help will be tremendously appreciated.