I am doing my first steps with freeswitch. I have installed on an Ubuntu 11.04 server: ii freeswitch 1.0.7~20110603-0natty6 ii freeswitch-codec-passthru-g7231 1.0.7~20110603-0natty6 ii freeswitch-codec-passthru-g729 1.0.7~20110603-0natty6 ii freeswitch-lang-de 1.0.7~20110603-0natty6 ii freeswitch-lang-en 1.0.7~20110603-0natty6 Following the tutorial at http://www.onlinesolution.co.nz/viewtopic.php?t=102 I modified /opt/freeswitch/conf/directory/default/1000.xml <include> <user id="1000"> <params> <param name="password" value="siptest"/> <param name="vm-password" value="siptest"/> </params> <variables> <variable name="toll_allow" value="domestic,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> Then I tried to connect to the server from another machine: +++ 27-6-2011 17:07:30.279796 INFO SIP ::send_sip_udp Send to: udp:192.168.2.100:5060 REGISTER sip:192.168.2.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.108;rport;branch=z9hG4bKcvkohckk Max-Forwards: 70 To: "FS Admin" <sip:1000@192.168.2.100> From: "FS Admin" <sip:1000@192.168.2.100>;tag=nyyvz Call-ID: zmjxppampyrrgxk@client.local.tld CSeq: 645 REGISTER Contact: <sip:1000@192.168.2.108>;expires=3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0 --- +++ 27-6-2011 17:07:30.281799 INFO SIP ::process_sip_msg Received from: udp:192.168.2.100:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.2.108;received=192.168.2.108;rport=5060;branch=z9hG4bKcvkohckk To: "FS Admin" <sip:1000@192.168.2.100>;tag=iairy From: "FS Admin" <sip:1000@192.168.2.100>;tag=nyyvz Call-ID: zmjxppampyrrgxk@client.local.tld CSeq: 645 REGISTER Server: Twinkle/1.4.2 Content-Length: 0 What could be wrong, so I can't connect? Al
Al, please, show us the content of the following files: conf/directory/default.xml conf/vars.xml May you also tell us if you see a message in fs_cli when you try to register ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 27/06/2011 à 22:05, Al Bogner a écrit :
I am doing my first steps with freeswitch.
I have installed on an Ubuntu 11.04 server:
ii freeswitch 1.0.7~20110603-0natty6 ii freeswitch-codec-passthru-g7231 1.0.7~20110603-0natty6 ii freeswitch-codec-passthru-g729 1.0.7~20110603-0natty6 ii freeswitch-lang-de 1.0.7~20110603-0natty6 ii freeswitch-lang-en 1.0.7~20110603-0natty6
Following the tutorial at http://www.onlinesolution.co.nz/viewtopic.php?t=102 I modified /opt/freeswitch/conf/directory/default/1000.xml
<include> <user id="1000"> <params> <param name="password" value="siptest"/> <param name="vm-password" value="siptest"/> </params> <variables> <variable name="toll_allow" value="domestic,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include>
Then I tried to connect to the server from another machine:
+++ 27-6-2011 17:07:30.279796 INFO SIP ::send_sip_udp Send to: udp:192.168.2.100:5060 REGISTER sip:192.168.2.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.108;rport;branch=z9hG4bKcvkohckk Max-Forwards: 70 To: "FS Admin" <sip:1000@192.168.2.100> From: "FS Admin" <sip:1000@192.168.2.100>;tag=nyyvz Call-ID: zmjxppampyrrgxk@client.local.tld CSeq: 645 REGISTER Contact: <sip:1000@192.168.2.108>;expires=3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0
---
+++ 27-6-2011 17:07:30.281799 INFO SIP ::process_sip_msg Received from: udp:192.168.2.100:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.2.108;received=192.168.2.108;rport=5060;branch=z9hG4bKcvkohckk To: "FS Admin" <sip:1000@192.168.2.100>;tag=iairy From: "FS Admin" <sip:1000@192.168.2.100>;tag=nyyvz Call-ID: zmjxppampyrrgxk@client.local.tld CSeq: 645 REGISTER Server: Twinkle/1.4.2 Content-Length: 0
What could be wrong, so I can't connect?
Al
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Am Di, 28 Jun 2011 01:55:08 CEST schrieb David Ponzone: Hi David,
please, show us the content of the following files: conf/directory/default.xml conf/vars.xml
These files are default. See below.
May you also tell us if you see a message in fs_cli when you try to register ?
/opt/freeswitch/bin/fs_cli Type /help <enter> to see a list of commands +OK log level [7] I see nothing else. Should I use a specific option? Do you want me to try another sipclient than twinkle? Linphone is installed too. Al cat /opt/freeswitch/conf/directory/default.xml <!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE FreeSWITCH works off the concept of users and domains just like email. You have users that are in domains for example 1000@domain.com. When freeswitch gets a register packet it looks for the user in the directory based on the from or to domain in the packet depending on how your sofia profile is configured. Out of the box the default domain will be the IP address of the machine running FreeSWITCH. This IP can be found by typing "sofia status" at the CLI. You will register your phones to the IP and not the hostname by default. If you wish to register using the domain please open vars.xml in the root conf directory and set the default domain to the hostname you desire. Then you would use the domain name in the client instead of the IP address to register with FreeSWITCH. NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE --> <include> <!--the domain or ip (the right hand side of the @ in the addr--> <domain name="$${domain}"> <params> <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> <variable name="default_gateway" value="$${default_provider}"/> <variable name="default_areacode" value="$${default_areacode}"/> <variable name="transfer_fallback_extension" value="operator"/> </variables> <groups> <group name="default"> <users> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </users> </group> <group name="sales"> <users> <!-- type="pointer" is a pointer so you can have the same user in multiple groups. It basically means to keep searching for the user in the directory. --> <user id="1000" type="pointer"/> <user id="1001" type="pointer"/> <user id="1002" type="pointer"/> <user id="1003" type="pointer"/> <user id="1004" type="pointer"/> </users> </group> <group name="billing"> <users> <user id="1005" type="pointer"/> <user id="1006" type="pointer"/> <user id="1007" type="pointer"/> <user id="1008" type="pointer"/> <user id="1009" type="pointer"/> </users> </group> <group name="support"> <users> <user id="1010" type="pointer"/> <user id="1011" type="pointer"/> <user id="1012" type="pointer"/> <user id="1013" type="pointer"/> <user id="1014" type="pointer"/> </users> </group> </groups> </domain> </include> cat /opt/freeswitch/conf/vars.xml <include> <!-- Preprocessor Variables These are introduced when configuration strings must be consistent across modules. NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead. WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any toll fraud in the future. It's your responsibility to secure your own system. This default config is used to demonstrate the feature set of FreeSWITCH. WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING --> <X-PRE-PROCESS cmd="set" data="default_password=1234"/> <!-- Did you change it yet? --> <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <!-- This setting is what sets the default domain FreeSWITCH will use if all else fails. FreeSWICH will default to $${local_ip_v4} unless changed. Changing this setting does affect the sip authentication. Please review conf/directory/default.xml for more information on this topic. --> <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/> <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/> <X-PRE-PROCESS cmd="set" data="use_profile=internal"/> <!-- Enable ZRTP globally you can override this on a per channel basis http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp) --> <X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/> <!-- Examples of codec options: (module must be compiled and loaded) codecname[@8000h|16000h|32000h[@XXi]] XX is the frame size must be multples allowed for the codec FreeSWITCH can support 10-120ms on some codecs. We do not support exceeding the MTU of the RTP packet. iLBC@30i - iLBC using mode=30 which will win in all cases. DVI4@8000h@20i - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10) DVI4@16000h@40i - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10) speex@8000h@20i - Speex 8kHz using 20ms ptime. speex@16000h@20i - Speex 16kHz using 20ms ptime. speex@32000h@20i - Speex 32kHz using 20ms ptime. BV16 - BroadVoice 16kb/s narrowband, 8kHz BV32 - BroadVoice 32kb/s wideband, 16kHz G7221@16000h - G722.1 16kHz (aka Siren 7) G7221@32000h - G722.1C 32kHz (aka Siren 14) CELT@32000h - CELT 32kHz, only 10ms supported CELT@48000h - CELT 48kHz, only 10ms supported GSM@40i - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms) G722 - G722 16kHz using default 20ms ptime. (multiples of 10) PCMU - G711 8kHz ulaw using default 20ms ptime. (multiples of 10) PCMA - G711 8kHz alaw using default 20ms ptime. (multiples of 10) G726-16 - G726 16kbit adpcm using default 20ms ptime. (multiples of 10) G726-24 - G726 24kbit adpcm using default 20ms ptime. (multiples of 10) G726-32 - G726 32kbit adpcm using default 20ms ptime. (multiples of 10) G726-40 - G726 40kbit adpcm using default 20ms ptime. (multiples of 10) AAL2-G726-16 - Same as G726-16 but using AAL2 packing. (multiples of 10) AAL2-G726-24 - Same as G726-24 but using AAL2 packing. (multiples of 10) AAL2-G726-32 - Same as G726-32 but using AAL2 packing. (multiples of 10) AAL2-G726-40 - Same as G726-40 but using AAL2 packing. (multiples of 10) LPC - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH) L16 - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly. These are the passthru audio codecs: G729 - G729 in passthru mode. (mod_g729) G723 - G723.1 in passthru mode. (mod_g723_1) AMR - AMR in passthru mode. (mod_amr) These are the passthru video codecs: (mod_h26x) H261 - H.261 Video H263 - H.263 Video H263-1998 - H.263-1998 Video H263-2000 - H.263-2000 Video H264 - H.264 Video RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for. 96 - AMR 97 - iLBC (30) 98 - iLBC (20) 99 - Speex 8kHz, 16kHz, 32kHz 100 - 101 - telephone-event 102 - 103 - 104 - 105 - 106 - BV16 107 - G722.1 (16kHz) 108 - 109 - 110 - 111 - 112 - 113 - 114 - CELT 32kHz, 48kHz 115 - G722.1C (32kHz) 116 - 117 - SILK 8kHz 118 - SILK 12kHz 119 - SILK 16kHz 120 - SILK 24kHz 121 - AAL2-G726-40 && G726-40 122 - AAL2-G726-32 && G726-32 123 - AAL2-G726-24 && G726-24 124 - AAL2-G726-16 && G726-16 125 - 126 - 127 - BV32 --> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> <!-- xmpp_client_profile and xmpp_server_profile xmpp_client_profile can be any string. xmpp_server_profile is appended to "dingaling_" to form the database name containing the "subscriptions" table. used by: dingaling.conf.xml enum.conf.xml --> <X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/> <X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/> <!-- THIS IS ONLY USED FOR DINGALING bind_server_ip Can be an ip address, a dns name, or "auto". This determines an ip address available on this host to bind. If you are separating RTP and SIP traffic, you will want to have use different addresses where this variable appears. Used by: dingaling.conf.xml --> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> <!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE If you're going to load test FreeSWITCH please input real IP addresses for external_rtp_ip and external_sip_ip --> <!-- external_rtp_ip Can be an one of: ip address: "12.34.56.78" a stun server lookup: "stun:stun.server.com" a DNS name: "host:host.server.com" where fs.mydomain.com is a DNS A record-useful when fs is on a dynamic IP address, and uses a dynamic DNS updater. If unspecified, the bind_server_ip value is used. Used by: sofia.conf.xml dingaling.conf.xml --> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> <!-- external_sip_ip Used as the public IP address for SDP. Can be an one of: ip address: "12.34.56.78" a stun server lookup: "stun:stun.server.com" a DNS name: "host:host.server.com" where fs.mydomain.com is a DNS A record-useful when fs is on a dynamic IP address, and uses a dynamic DNS updater. If unspecified, the bind_server_ip value is used. Used by: sofia.conf.xml dingaling.conf.xml --> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/> <!-- unroll-loops Used to turn on sip loopback unrolling. --> <X-PRE-PROCESS cmd="set" data="unroll_loops=true"/> <!-- outbound_caller_id and outbound_caller_name The caller ID telephone number we should use when calling out. Used by: conference.conf.xml and user directory for default outbound callerid name and number. --> <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/> <!-- various debug and defaults --> <X-PRE-PROCESS cmd="set" data="call_debug=false"/> <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/> <X-PRE-PROCESS cmd="set" data="default_areacode=918"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="be-ring=%(1000,3000,425)"/> <X-PRE-PROCESS cmd="set" data="ca-ring=%(2000,4000,440,480)"/> <X-PRE-PROCESS cmd="set" data="cn-ring=%(1000,4000,450)"/> <X-PRE-PROCESS cmd="set" data="cy-ring=%(1500,3000,425)"/> <X-PRE-PROCESS cmd="set" data="cz-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="de-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="dk-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="dz-ring=%(1500,3500,425)"/> <X-PRE-PROCESS cmd="set" data="eg-ring=%(2000,1000,475,375)"/> <X-PRE-PROCESS cmd="set" data="fi-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440)"/> <X-PRE-PROCESS cmd="set" data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/> <X-PRE-PROCESS cmd="set" data="hu-ring=%(1250,3750,425)"/> <X-PRE-PROCESS cmd="set" data="il-ring=%(1000,3000,400)"/> <X-PRE-PROCESS cmd="set" data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/> <X-PRE-PROCESS cmd="set" data="jp-ring=%(1000,2000,420,380)"/> <X-PRE-PROCESS cmd="set" data="ko-ring=%(1000,2000,440,480)"/> <X-PRE-PROCESS cmd="set" data="pk-ring=%(1000,2000,400)"/> <X-PRE-PROCESS cmd="set" data="pl-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="ro-ring=%(1850,4150,475,425)"/> <X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425)"/> <X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425)"/> <X-PRE-PROCESS cmd="set" data="sa-ring=%(1200,4600,425)"/> <X-PRE-PROCESS cmd="set" data="tr-ring=%(2000,4000,450)"/> <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/> <X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440,480)"/> <X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/> <X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> <!-- Setting up your default sip provider is easy. Below are some values that should work in most cases. These are for conf/directory/default/example.com.xml --> <X-PRE-PROCESS cmd="set" data="default_provider=example.com"/> <X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/> <X-PRE-PROCESS cmd="set" data="default_provider_password=password"/> <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/> <!-- true or false --> <X-PRE-PROCESS cmd="set" data="default_provider_register=false"/> <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/> <!-- SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls --> <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/> <!-- Internal SIP Profile --> <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/> <!-- External SIP Profile --> <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/> </include>
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 27/06/2011 à 22:05, Al Bogner a écrit :
I am doing my first steps with freeswitch.
I have installed on an Ubuntu 11.04 server:
ii freeswitch 1.0.7~20110603-0natty6 ii freeswitch-codec-passthru-g7231 1.0.7~20110603-0natty6 ii freeswitch-codec-passthru-g729 1.0.7~20110603-0natty6 ii freeswitch-lang-de 1.0.7~20110603-0natty6 ii freeswitch-lang-en 1.0.7~20110603-0natty6
Following the tutorial at http://www.onlinesolution.co.nz/viewtopic.php?t=102 I modified /opt/freeswitch/conf/directory/default/1000.xml
<include> <user id="1000"> <params> <param name="password" value="siptest"/> <param name="vm-password" value="siptest"/> </params> <variables> <variable name="toll_allow" value="domestic,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include>
Then I tried to connect to the server from another machine:
+++ 27-6-2011 17:07:30.279796 INFO SIP ::send_sip_udp Send to: udp:192.168.2.100:5060 REGISTER sip:192.168.2.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.108;rport;branch=z9hG4bKcvkohckk Max-Forwards: 70 To: "FS Admin" <sip:1000@192.168.2.100> From: "FS Admin" <sip:1000@192.168.2.100>;tag=nyyvz Call-ID: zmjxppampyrrgxk@client.local.tld CSeq: 645 REGISTER Contact: <sip:1000@192.168.2.108>;expires=3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0
---
+++ 27-6-2011 17:07:30.281799 INFO SIP ::process_sip_msg Received from: udp:192.168.2.100:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.2.108;received=192.168.2.108;rport=5060;branch=z9hG4bKcvkohckk To: "FS Admin" <sip:1000@192.168.2.100>;tag=iairy From: "FS Admin" <sip:1000@192.168.2.100>;tag=nyyvz Call-ID: zmjxppampyrrgxk@client.local.tld CSeq: 645 REGISTER Server: Twinkle/1.4.2 Content-Length: 0
What could be wrong, so I can't connect?
Al
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yes, please, try another one, juste in case. For instance, Zoiper works fine. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 28/06/2011 à 12:42, Al Bogner a écrit :
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participants (2)
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Al Bogner
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David Ponzone