I want to register a telephone 2010 (a remote extension) to my FS server. My server has dynamic ip address, registered with DynDNS. (dyndns ip address is: xxxx.ftpaccess.cc) In the remote telephone I programmed: a. stun server=stun.freeswitch.org b. sip port=5090 (i replaced all the 5060 with 5090 at several places) c. rtp port=17000 d. domain= xxxx.ftpaccess.cc (my dyndns) The remote phone is also on a dynamic ip address. I followed the instructions on http://wiki.freeswitch.org/wiki/Example_Offsite_phones Step 1. I created an extension 2010 in /..../directory/default.xml ================================================= <include> user id="2010" mailbox="2010"> params> param name="password" value="2010"/> param name="vm-password" value="2010"/> /params> variables> variable name="accountcode" value="2010"/> variable name="user_context" value="default"/> variable name="effective_caller_id_name" value="Terje"/> variable name="effective_caller_id_number" value="2010"/> /variables> /user> </include> Setp 2: I created a new profile. called "doublenat" profile set to port 5090 in /..../sip_profiles/doublenat.xml ================================================================================== <profile name="doublenat"> gateways> X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/> /gateways> settings> param name="debug" value="0"/> param name="sip-trace" value="no"/> param name="rfc2833-pt" value="101"/> param name="sip-port" value="5090"/> param name="dialplan" value="XML"/> param name="context" value="public"/> param name="dtmf-duration" value="100"/> param name="codec-prefs" value="$${outbound_codec_prefs}"/> param name="use-rtp-timer" value="true"/> param name="hold-music" value="$${hold_music}"/> param name="rtp-timer-name" value="soft"/> param name="manage-presence" value="false"/> param name="aggressive-nat-detection" value="true"/> param name="apply-nat-acl" value="rfc1918"/> param name="inbound-codec-negotiation" value="generous"/> param name="nonce-ttl" value="60"/> param name="auth-calls" value="false"/> param name="rtp-timeout-sec" value="1800"/> param name="rtp-ip" value="$${local_ip_v4}"/> param name="sip-ip" value="$${local_ip_v4}"/> param name="ext-rtp-ip" value="$${external_rtp_ip}"/> param name="ext-sip-ip" value="$${external_sip_ip}"/> param name="rtp-timeout-sec" value="300"/> param name="rtp-hold-timeout-sec" value="1800"/> /settings> </profile> Step 3: Inside the /..../conf/dialplan/public.xml, I put ========================================= <extension name="public_extensions"> <condition field="destination_number" expression="^(20[01][0-9])$"> <action application="bridge" data="sofia/doublenat/$1%$${domain} "/> </condition> </extension> However the phone can't register. In the FS CLI window I get: 2008-09-24 20:10:39 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [2010@xxxx.ftpaccess.cc] You must define a domain called 'henkoegema.ftpaccess.cc' in your directory and add a user with the id="2010" attribute and you must configure your device to use the proper domain in it's authentication credentials. %-| This line repeats and repeats and repeats. Evenso the warning is clear, I'm not sure what exactly I should do. (and where) :confused: Henk -- View this message in context: http://www.nabble.com/Registering-remote-extensions.-tp19655521p19655521.htm... Sent from the Freeswitch-users mailing list archive at Nabble.com.
I still haven't been able to solve my problem. Still getting: ========= 2008-09-26 11:02:59 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [2010@xxxx.ftpaccess.cc] You must define a domain called 'xxxx.ftpaccess.cc' in your directory and add a user with the id="2010" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2008-09-26 11:03:37 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [2010@xxxx.ftpaccess.cc] You must define a domain called 'xxxx.ftpaccess.cc' in your directory and add a user with the id="2010" attribute and you must configure your device to use the proper domain in it's authentication credentials. ...............etc.....etc....etc Have no idea how to solve this. :rules: Henk -- View this message in context: http://www.nabble.com/Registering-remote-extensions.-tp19655521p19684994.htm... Sent from the Freeswitch-users mailing list archive at Nabble.com.
Doesn't the long winded error message even give you a hint? I am quite suprised by how much difficulty people seem to have with this concept. SIP and FS are both domain based. Just like email, IM and most other internet protocols. If you are pointing a sip client at FS, you need to specify the same domain in your sip phone that you have FS configured to be in charge of. It's like if you wanted to go to our website you must point your browser at freeswitch.org or you will not get there. The IP that leads to freeswitch.org may lead to 200 websites and the domain name is the only realy way to tell where you want to go. If you just tell the sip client the address of FS and FS is set to manage a specific IP or domain name it will not be able to find the user. like the error says, edit your directory and make sure the user and domain in your directory config and the user and domain in your sip client match, that's all that's to it. On Fri, Sep 26, 2008 at 4:12 AM, henkoegema <pabx_freeswitch@telenet.be>wrote:
I still haven't been able to solve my problem.
Still getting: ========= 2008-09-26 11:02:59 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [2010@xxxx.ftpaccess.cc] You must define a domain called 'xxxx.ftpaccess.cc' in your directory and add a user with the id="2010" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2008-09-26 11:03:37 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [2010@xxxx.ftpaccess.cc] You must define a domain called 'xxxx.ftpaccess.cc' in your directory and add a user with the id="2010" attribute and you must configure your device to use the proper domain in it's authentication credentials. ...............etc.....etc....etc
Have no idea how to solve this. :rules:
Henk
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony Minessale-2 wrote:
Doesn't the long winded error message even give you a hint?
Yes, it did. But I couldn't find the right place and syntax Anthony Minessale-2 wrote:
I am quite suprised by how much difficulty people seem to have with this concept.
May be that's because I am one (or two) generations older than you are. :handshake: When I started working, I still had to wait for 30 years before the first (home) computer came in the marked. We were still working with analogue PBX's, and after a few years we started with the first SPC (Stored Program Control) PBX. What we now call digital PBX. I'm still in the process of learning XML, Perl and Python. I'm doing all this because I think FS is a wonderful piece of engineering. Anthony Minessale-2 wrote:
SIP and FS are both domain based. Just like email, IM and most other internet protocols.
If you are pointing a sip client at FS, you need to specify the same domain in your sip phone that you have FS configured to be in charge of. It's like if you wanted to go to our website you must point your browser at freeswitch.org or you will not get there. The IP that leads to freeswitch.org may lead to 200 websites and the domain name is the only realy way to tell where you want to go. If you just tell the sip client the address of FS and FS is set to manage a specific IP or domain name it will not be able to find the user.
Thanks for your explanation. =) Anthony Minessale-2 wrote:
like the error says, edit your directory and make sure the user and domain in your directory config and the user and domain in your sip client match, that's all that's to it.
What I did is following (may be it's useful to other people ?) 1. in the file ..../conf/directory/default.xml I changed the line: <domain name="$${domain}"> to <domain name="xxxx.ftpaccess.cc"> 2. in each remote extension I added (see bold lines) <include> <user id="2010" mailbox="2010"> <domain name="xxxx.ftpaccess.cc"> <params> param name="password" value="2010"/> param name="vm-password" value="2010"/> </params> <variables> <variable name="accountcode" value="2010"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Terje"/> <variable name="effective_caller_id_number" value="2010"/> </variables> </domain> </user> </include> I now have three remote extensions: Two extensions are registered in the profile internal: freeswitch@ubuntu> sofia status profile internal Registrations: ================================================================================================= Call-ID 1f491145-b7274a7c@193.173.175.2xx User 2011@xxxx.ftpaccess.cc Contact 2011 <sip:2011@193.173.175.2xx:5090> <-----------FXS line Agent Linksys/SPA3000-3.1.18(GW) Status Registered(UDP)(unknown) EXP(2008-09-27 12:31:56) Call-ID a694c000-2ff772fd@193.173.175.2xx User 2012@xxxx.ftpaccess.cc Contact 2012 <sip:2012@193.173.175.2xx:5091> <-------------FXO line Agent Linksys/SPA3000-3.1.18(GW) Status Registered(UDP)(unknown) EXP(2008-09-27 13:11:25) and one extension is registered in the profile doublenat: freeswitch@ubuntu> sofia status profile doublenat Registrations: ================================================================================================= Call-ID 1554239560@192_168_0_197 User 2010@xxxx.ftpaccess.cc Contact Terje <sip:2010@79.160.18.xx:5090> Agent TARGA IP FON 1020 Status Registered(UDP)(unknown) EXP(2008-09-27 11:30:20) ================================================================================================= I assume(??) the reason that they are in different profiles is that ext. 2010 is behind a remote nat and 2011 and 2012 are not. I have come so far now that my three remote extensions are registered with my FS server. The only outstanding issue is that if I call each of them I get: 2010 (2011, 2012) is not available. Record your message at the tone....... :working: Henk -- View this message in context: http://www.nabble.com/Registering-remote-extensions.-tp19655521p19701306.htm... Sent from the Freeswitch-users mailing list archive at Nabble.com.
henkoegema wrote:
The only outstanding issue is that if I call each of them I get:
2010 (2011, 2012) is not available. Record your message at the tone.......
Forget to include debug in previous message: freeswitch@ubuntu> 2008-09-27 17:27:04 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/2000@192.168.2.11 [bd05f8ee-8ca8-11dd-b9c2-9d3c2758233a] 2008-09-27 17:27:04 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 2000->2010@default 2008-09-27 17:27:04 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2008-09-27 17:27:04 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/2000.2008-09-27-17-27-04.wav 2008-09-27 17:27:04 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-09-27 17:27:04 [WARNING] switch_ivr_originate.c:578 switch_ivr_originate() No origination URL specified! 2008-09-27 17:27:04 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [DESTINATION_OUT_OF_ORDER] 2008-09-27 17:27:04 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2008-09-27 17:27:04 [NOTICE] mod_dptools.c:601 answer_function() Channel [sofia/internal/2000@192.168.2.11] has been answered -- View this message in context: http://www.nabble.com/Registering-remote-extensions.-tp19655521p19703715.htm... Sent from the Freeswitch-users mailing list archive at Nabble.com.
Anthony Minessale-2 wrote:
I am quite suprised by how much difficulty people seem to have with this concept.
I am talking about the concept of user@domain.com. The point is not to insult you it's to find out what the difficult to understand thing is so we can fix it. The error message has been tweaked many times to try and explain the problem to the user, do you have any suggestions? What year are we talking about btw, the apple I which is not even the first computer was released in 1976 so were you really doing telephony stuff in the 40's ? Anyway, you should back up your config and upgrade to the latest SVN trunk with fresh install and try again. I can see by those error messages you have a much older revision of the default config. Just try the fresh vanilla install install, you should be able to register any 2 phones at an extension of 1001 and 1002 with password 1234, use the ip of your server as the domain. On Sat, Sep 27, 2008 at 4:48 AM, henkoegema <pabx_freeswitch@telenet.be>wrote:
Anthony Minessale-2 wrote:
Doesn't the long winded error message even give you a hint?
Yes, it did. But I couldn't find the right place and syntax
Anthony Minessale-2 wrote:
I am quite suprised by how much difficulty people seem to have with this concept.
May be that's because I am one (or two) generations older than you are. :handshake: When I started working, I still had to wait for 30 years before the first (home) computer came in the marked. We were still working with analogue PBX's, and after a few years we started with the first SPC (Stored Program Control) PBX. What we now call digital PBX. I'm still in the process of learning XML, Perl and Python. I'm doing all this because I think FS is a wonderful piece of engineering.
Anthony Minessale-2 wrote:
SIP and FS are both domain based. Just like email, IM and most other internet protocols.
If you are pointing a sip client at FS, you need to specify the same domain in your sip phone that you have FS configured to be in charge of. It's like if you wanted to go to our website you must point your browser at freeswitch.org or you will not get there. The IP that leads to freeswitch.org may lead to 200 websites and the domain name is the only realy way to tell where you want to go. If you just tell the sip client the address of FS and FS is set to manage a specific IP or domain name it will not be able to find the user.
Thanks for your explanation. =)
Anthony Minessale-2 wrote:
like the error says, edit your directory and make sure the user and
domain
in your directory config and the user and domain in your sip client match, that's all that's to it.
What I did is following (may be it's useful to other people ?)
1. in the file ..../conf/directory/default.xml I changed the line: <domain name="$${domain}"> to <domain name="xxxx.ftpaccess.cc">
2. in each remote extension I added (see bold lines)
<include> <user id="2010" mailbox="2010"> <domain name="xxxx.ftpaccess.cc"> <params> param name="password" value="2010"/> param name="vm-password" value="2010"/> </params> <variables> <variable name="accountcode" value="2010"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Terje"/> <variable name="effective_caller_id_number" value="2010"/> </variables> </domain> </user> </include>
I now have three remote extensions:
Two extensions are registered in the profile internal: freeswitch@ubuntu> sofia status profile internal Registrations:
================================================================================================= Call-ID 1f491145-b7274a7c@193.173.175.2xx User 2011@xxxx.ftpaccess.cc Contact 2011 <sip:2011@193.173.175.2xx:5090> <-----------FXS line Agent Linksys/SPA3000-3.1.18(GW) Status Registered(UDP)(unknown) EXP(2008-09-27 12:31:56)
Call-ID a694c000-2ff772fd@193.173.175.2xx User 2012@xxxx.ftpaccess.cc Contact 2012 <sip:2012@193.173.175.2xx:5091> <-------------FXO line Agent Linksys/SPA3000-3.1.18(GW) Status Registered(UDP)(unknown) EXP(2008-09-27 13:11:25)
and one extension is registered in the profile doublenat: freeswitch@ubuntu> sofia status profile doublenat Registrations:
================================================================================================= Call-ID 1554239560@192_168_0_197 User 2010@xxxx.ftpaccess.cc Contact Terje <sip:2010@79.160.18.xx:5090> Agent TARGA IP FON 1020 Status Registered(UDP)(unknown) EXP(2008-09-27 11:30:20)
=================================================================================================
I assume(??) the reason that they are in different profiles is that ext. 2010 is behind a remote nat and 2011 and 2012 are not.
I have come so far now that my three remote extensions are registered with my FS server.
The only outstanding issue is that if I call each of them I get:
2010 (2011, 2012) is not available. Record your message at the tone....... :working:
Henk
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony Minessale-2 wrote:
Anthony Minessale-2 wrote:
Just try the fresh vanilla install install, ......
freeswitch@ubuntu> version FreeSWITCH Version 1.0.trunk (9687) <------ Is this the latest one ?? Henk -- View this message in context: http://www.nabble.com/Registering-remote-extensions.-tp19655521p19706532.htm... Sent from the Freeswitch-users mailing list archive at Nabble.com.
close enough, more importantly let it install all the defaults again. mv /usr/local/freeswitch /usr/local/freeswitch.bak and make install again to get the latest defaults On Sat, Sep 27, 2008 at 4:23 PM, henkoegema <pabx_freeswitch@telenet.be>wrote:
Anthony Minessale-2 wrote:
Anthony Minessale-2 wrote:
Just try the fresh vanilla install install, ......
freeswitch@ubuntu> version FreeSWITCH Version 1.0.trunk (9687) <------ Is this the latest one ??
Henk
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony Minessale-2 wrote:
close enough, more importantly let it install all the defaults again.
mv /usr/local/freeswitch /usr/local/freeswitch.bak
and make install again to get the latest defaults
Don't I have the latest defaults when I do: 1. rm -rf /usr/local/freeswitch and 2. svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Henk -- View this message in context: http://www.nabble.com/Registering-remote-extensions.-tp19655521p19706816.htm... Sent from the Freeswitch-users mailing list archive at Nabble.com.
yes On Sat, Sep 27, 2008 at 5:01 PM, henkoegema <pabx_freeswitch@telenet.be>wrote:
Anthony Minessale-2 wrote:
close enough, more importantly let it install all the defaults again.
mv /usr/local/freeswitch /usr/local/freeswitch.bak
and make install again to get the latest defaults
Don't I have the latest defaults when I do: 1. rm -rf /usr/local/freeswitch and 2. svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
Henk
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
assuming you are not checking out to the same dir you installed it too. and assuming you are not repeating that checkout command over and existing install. once you check out the trunk you should do subsequent upgrades with "make current" On Sat, Sep 27, 2008 at 5:01 PM, henkoegema <pabx_freeswitch@telenet.be>wrote:
Anthony Minessale-2 wrote:
close enough, more importantly let it install all the defaults again.
mv /usr/local/freeswitch /usr/local/freeswitch.bak
and make install again to get the latest defaults
Don't I have the latest defaults when I do: 1. rm -rf /usr/local/freeswitch and 2. svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch
Henk
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (2)
-
Anthony Minessale
-
henkoegema