Re: [Freeswitch-users] Affordable GSM gateway for one cellphone?
Fred-145 wrote: > > Hello > > I'm selling a basic solution for SOHO customers (FS is installed on their > work computer running Windows or Macs) to handle an analog phone line. > When they're on the road, in addition or instead of getting a notification > by e-mail when someone calls their office, some users might want to have > the Freeswitch server actually ring their cellphone so they can take > calls. > > Besides taking a subscription with a VoIP provider that the Freeswitch > server will use to ring their cellphone, I'd like to know what my options > are when it comes to setting up a GSM gateway on the customer's premises, > in case they don't want to depend on the Internet. > > Are there Freeswitch-compatible, affordable solutions to handle a single > GSM subscription? I guess all it takes is having them take a second > subscription with their GSM provider and inserting the SIM chip inside the > gateway to have Freeswitch ring their cellphone, but I've never used those > things. > > Thank you. > I have been using http://www.portech.com.tw/p3-product1_1.asp?Pid=13 for years with Asterisk and Freeswitch. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530400.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Different files (?)
What is the difference between the following 3 files: 1. freeswitch-1.0.3.tar.gz19-Feb-2009 00:34 26M 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:0022M 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:0249M Which one should I download, if I want the latest (newest) ? -- View this message in context: http://www.nabble.com/Different-files-%28-%29-tp22670515p22670515.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] GSM gateway MV370
For solution see: http://www.nabble.com/Dialplan-translation-to-FS.-ts18983597.html -- View this message in context: http://www.nabble.com/GSM-gateway-MV370-tp18932091p19006591.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] any VOIP service provider support the freeswitch now?
http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing Tested_Phone_Providers_Listing -- View this message in context: http://www.nabble.com/any-VOIP-service-provider-support-the-freeswitch-now--tp19337344p19337894.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to divert a virtual PSTN line to another server ?
I'm still looking for a solution. :working: henkoegema wrote: > > I use a 'virtual' PSTN line (voip trunk) from (http://www.voxbone.com) as > incoming external line to my Asterisk server (192.168.1.100) > > In my router I have have : > Application Start End ProtocolIP Address > - > SIP 5004 to5082 Both(UDP&TCP) 192.168.1.100 > RTP 5090 to 5100UDP > 192.168.1.100 > > > That works OK. > > > Now I want to divert that PSTN line from Asterisk to my Freeswitch server > (192.168.1.101) > So I changed in my router the ip addreese from 192.168.1.100 to > 192.168.1.101 > > Application Start End ProtocolIP Address > - > SIP 5004 to5082 Both(UDP&TCP) 192.168.1.101 > RTP 5090 to 5100UDP > 192.168.1.101 > > > But.when an external call comes in, it still goes to Asterisk. > > Am I on the wrong track or ... (?) > > Rgds > Henk > > > > > ___ > 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 > > -- View this message in context: http://www.nabble.com/How-to-divert-a-virtual-PSTN-line-to-another-server---tp19335394p19366567.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to divert a virtual PSTN line to another server ?
I have not experienced any problem with port-forwarding. Is my problem a port-forwarding problem or could it be something else. :confused: Robert Dyck wrote: > > If your port forwarding is not working perhaps you should seek technical > support from the manufacturer of your particular router. Is there a forum > for > this router? > -- View this message in context: http://www.nabble.com/How-to-divert-a-virtual-PSTN-line-to-another-server---tp19335394p19369161.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to divert a virtual PSTN line to another server ?
It's getting more mysterious to me.:confused: If have flipped Asterisk and FS. Asterisk is now at 192.168.1.101 and FS is now at 192.168.1.100 Port-forwarding is: SIP 5004-5082 UDP/TCP 192.168.1.100 But incoming calls STILL go to Asterisk. :confused: Which it shoudn't. When I shutdown Asterisk, I get not-reachable (?) tone. (from my mobile) Henk Ivan C Myrvold wrote: > > If your call ends up at the IP address where Asterisk is running, > clearly your port forwarding is not working. Try to flip the ip > addresses of the Asterisk and FreeSWITCH, and see if it now ends up at > the FreeSWITCH. > > Ivan > > Den 8. sep.. 2008 kl. 12:03 skrev henkoegema: > -- View this message in context: http://www.nabble.com/How-to-divert-a-virtual-PSTN-line-to-another-server---tp19335394p19375549.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to divert a virtual PSTN line to another server ?
To see what's happening with the voip packets from my router I want to do following: I have Wireshark installed on pc 192.168.1.101. Can (if so, how?) I monitor traffic from my router (192.168.1.1) and flowing into the LAN network (f.e. to 192.168.1.100) ? Or can I only monitor traffic from and to 192.168.101 (where Wireshark is installed) ? Henk -- View this message in context: http://www.nabble.com/How-to-divert-a-virtual-PSTN-line-to-another-server---tp19335394p19408518.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Can't find user (Anonymous calls)
When I make a call (from SJphone) to sip:[EMAIL PROTECTED] than I can see in Freeswitch: 2008-09-18 10:16:19 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [EMAIL PROTECTED] 2008-09-18 10:16:19 [DEBUG] sofia_reg.c:553 sofia_reg_handle_register() send challange for [EMAIL PROTECTED] How to I answer this call for [EMAIL PROTECTED] at extension 2000 ? (also calling 2000 (which is a registered number in FS) gives me: 2008-09-18 10:25:39 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [EMAIL PROTECTED] 2008-09-18 10:25:39 [DEBUG] sofia_reg.c:553 sofia_reg_handle_register() send challange for [EMAIL PROTECTED] ) Henk -- View this message in context: http://www.nabble.com/Can%27t-find-user-%28Anonymous-calls%29-tp19548337p19548337.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Can't find user (Anonymous calls)
Anthony Minessale-2 wrote: > > Turn off the auth-calls = true in your profile if you do not require > authentication for the inbound calls. or add accounts > for your phone so it can make authenticated calls. > In which file do I need to do that ? (Have gone through most of them, but can't find it):confused: Anthony Minessale-2 wrote: > > btw, you appear to be using an older version of the code. You may want to > update to help us beta test the 1.0.2 release. > I will. :-) -- View this message in context: http://www.nabble.com/Can%27t-find-user-%28Anonymous-calls%29-tp19548337p19554755.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Can't find user (Anonymous calls)
Anthony Minessale-2 wrote: > > in the sofia profile > > auth-calls is set to true > and context is set to public > > change auth-calls to false > and change context to default > Changed the above settings in ./conf/sip-profiles/internal.xml and now ..:drunk: Henk -- View this message in context: http://www.nabble.com/Can%27t-find-user-%28Anonymous-calls%29-tp19548337p19558659.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] How to upgrade ?
At the moment I'm using FS version 1.0.0. I want to upgrade to the latest version. Can I just type #make current(?):working: without loosing my own (or edited standard) conf files ? -- View this message in context: http://www.nabble.com/How-to-upgrade---tp19589317p19589317.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to upgrade ?
I downloaded from http://wiki.freeswitch.org/wiki/Download_Freeswitch Latest Snapshot Which version is this : [EMAIL PROTECTED]> version FreeSWITCH Version 1.0.trunk (9609) :confused: [EMAIL PROTECTED]> Is this the same as 1.0.2 (the latest ?) Henk -- View this message in context: http://www.nabble.com/How-to-upgrade---tp19589317p19602923.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Some common extensions for testing
Just installed FS 1.0.1 When testing extension (calling from extension 2000) it rings 3 times and than I get busy tone. [EMAIL PROTECTED]> 2008-09-22 11:02:10 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [240f78bc-8885-11dd-8238-95905a8f75b0] 2008-09-22 11:02:10 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 2000->[EMAIL PROTECTED] 2008-09-22 11:02:10 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/2001 [24106614-8885-11dd-8238-95905a8f75b0] 2008-09-22 11:02:11 [NOTICE] sofia.c:2167 sofia_handle_sip_i_state() Ring-Ready sofia/internal/2001! 2008-09-22 11:02:11 [NOTICE] mod_sofia.c:1058 sofia_receive_message() Ring-Ready sofia/internal/[EMAIL PROTECTED] 2008-09-22 11:02:11 [NOTICE] switch_ivr_originate.c:1148 switch_ivr_originate() Ring Ready sofia/internal/[EMAIL PROTECTED] 2008-09-22 11:02:26 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/2001 [CS_CONSUME_MEDIA] [USER_BUSY] 2008-09-22 11:02:26 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: USER_BUSY:-/ 2008-09-22 11:02:26 [NOTICE] mod_dptools.c:1816 audio_bridge_function() Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [USER_BUSY] 2008-09-22 11:02:26 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 1 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-09-22 11:02:26 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] 2008-09-22 11:02:26 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 2 (sofia/internal/2001) Ended 2008-09-22 11:02:26 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/2001 [CS_HANGUP] Calling 9996 - standard echo test , it rings 10 times and than disconnects. 2008-09-22 11:07:04 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NO_ANSWER Everything was working fine initially in version 1.0.0. Then all of a sudden, I got the problems (mentioned above) in version 1.0.0, so I decided to install a new version (1.0.1) Unfortunately the problem persists. When calling 5000 (demo ivr), option 1 (conference) OK option 2 (echo test) NOK option 3 (moh) NOK option 4 (sample ivr submenu) OK option 5 (screaming monkey) NOK When calling internally from extension 2002 (SJPhone) to 2000 (Analog phone with Linksys SPA 3000) the moh is very cracky. Doing the same test via an external DID line gives a fine moh ! Henk -- View this message in context: http://www.nabble.com/Some-common-extensions-for-testing-tp19604479p19604479.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Some common extensions for testing
henkoegema wrote: > > Just installed FS 1.0.1 > > > > 2008-09-22 11:02:26 [INFO] mod_dptools.c:1789 audio_bridge_function() > Originate Failed. Cause: USER_BUSY:-/ > The problem was caused by myself :blush: My apology. I had defined numbers starting with 9, as calls to my GSM gateway. After changing that prefix to another number the problem(s) was solved. Henk -- View this message in context: http://www.nabble.com/Some-common-extensions-for-testing-tp19604479p19610111.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Problems with configure script
Jon Bruel wrote: > > I have got the trunk version of FreeSWITCH > What is the difference of the trunk version compared to version 1.0.x ? -- View this message in context: http://www.nabble.com/Problems-with-configure-script-tp19621504p19622583.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Re cording info
What is the purpose of these lines: :-/ ? == Only this file info I see in .../recordings 2008-09-23-19-35-15_henk_324763788xx.wav 2008-09-23-19-37-56_henk_324763788xx.wav as a result of: Henk -- View this message in context: http://www.nabble.com/Recording-info-tp19633360p19633360.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Re cording info
Michael Collins-8 wrote: > > Many audio formats let you embed meta data with this kind of > information. That's how you can hover your mouse over a wav file in > Windows and it pops up a little box that says, "Artist: Britney Spears, > Title: Hit Me Baby One More Time, etc." > -MC > I have "Windows Media Player" and "VLC media player". VLC media player even has an option to show Media Informatie, but I don't seen any meta data. Do you have an example of a program that does show embed meta data. (in Linux or (Windows) ) :-D -- View this message in context: http://www.nabble.com/Recording-info-tp19633360p19635776.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Re gistering remote extensions.
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: .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= .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
=
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>
Setp 2: I created a new profile. called "doublenat" profile set to port
5090 in //sip_profiles/doublenat.xml
==
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>
Step 3: Inside the //conf/dialplan/public.xml, I put
=
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 [EMAIL PROTECTED]
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
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Re: [Freeswitch-users] Re gistering remote extensions.
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 [EMAIL PROTECTED] You must define a domain called '.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 [EMAIL PROTECTED] You must define a domain called '.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.etcetc Have no idea how to solve this. :rules: Henk -- View this message in context: http://www.nabble.com/Registering-remote-extensions.-tp19655521p19684994.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Re gistering remote extensions.
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: to 2. in each remote extension I added (see bold lines) param name="password" value="2010"/> param name="vm-password" value="2010"/> I now have three remote extensions: Two extensions are registered in the profile internal: [EMAIL PROTECTED]> sofia status profile internal Registrations: = Call-ID [EMAIL PROTECTED] User[EMAIL PROTECTED] Contact 2011 <---FXS line Agent Linksys/SPA3000-3.1.18(GW) Status Registered(UDP)(unknown) EXP(2008-09-27 12:31:56) Call-ID [EMAIL PROTECTED] User[EMAIL PROTECTED] Contact 2012 <-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: [EMAIL PROTECTED]> sofia status profile doublenat Registrations: = Call-ID [EMAIL PROTECTED] User[EMAIL PROTECTED] Contact Terje 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.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Re gistering remote extensions.
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: [EMAIL PROTECTED]> 2008-09-27 17:27:04 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [bd05f8ee-8ca8-11dd-b9c2-9d3c2758233a] 2008-09-27 17:27:04 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 2000->[EMAIL PROTECTED] 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/[EMAIL PROTECTED] has been answered -- View this message in context: http://www.nabble.com/Registering-remote-extensions.-tp19655521p19703715.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Re gistering remote extensions.
Anthony Minessale-2 wrote: > > Anthony Minessale-2 wrote: > > Just try the fresh vanilla install install, .. > > [EMAIL PROTECTED]> 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.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Re gistering remote extensions.
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.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] No ring-back tone.
When I dial with my mobile phone to my DID line, I don't get a ring-back tone. :confused: The extension where the DID line is answered is like this: I tried to add after (and before) the 'bridge' application, but that doesn't help. Henk -- View this message in context: http://www.nabble.com/No-ring-back-tone.-tp19767182p19767182.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] No ring-back tone.
Brian West-3 wrote: > > If the call isn't answered you use ringback .. if the call is already > answered its transfer_ringback. > After adding this line: the ringback was working. =) -- View this message in context: http://www.nabble.com/No-ring-back-tone.-tp19767182p19768257.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] No ring-back tone.
Brian West-3 wrote: > > Was? or "is" working? :P > is=^D -- View this message in context: http://www.nabble.com/No-ring-back-tone.-tp19767182p19769648.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] GotoIf
How do I do this in FS ? :
exten => s,n,GotoIf($[${CALLERID(num)}=32484120169]?default,1000,1)
exten => s,n,GotoIf($[${CALLERID(num)}=4732803429]?default,terje,1)
exten => s,n,GotoIf($[${CALLERID(num)}=32476378561]?henk_gsm_disa,s,1)
The Asterisk to Freeswitch Rosetta stone says:
GotoIf Conditions in dialplan (
but I can't figure out the right syntax.
Henk
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Re: [Freeswitch-users] GotoIf
henkoegema wrote:
>
> How do I do this in FS ? :
> exten => s,n,GotoIf($[${CALLERID(num)}=32484120169]?default,1000,1)
> exten => s,n,GotoIf($[${CALLERID(num)}=4732803429]?default,terje,1)
> exten => s,n,GotoIf($[${CALLERID(num)}=32476378561]?henk_gsm_disa,s,1)
>
>
> The Asterisk to Freeswitch Rosetta stone says:
> GotoIfConditions in dialplan ( expression="foo">
> but I can't figure out the right syntax.
>
> Henk
>
I've made some progress in the mean time.
I made a test extension 2020:
:clap:
When I call 2020 from 2000, I'l get the MOH.
When I swap 2000 and 2002 I will NOT get MOH :
:confused:
Henk
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Re: [Freeswitch-users] GotoIf
Brian West-3 wrote: > > Actually try this: > > > > > > > > > > > > > > > > > > > > . > . > The above works. :jumping: === But I'm still puzzling with this: When I call 2020 from 2000 it's ok. When I call 2020 from 2002 i't NOT ok. (Call failed: 404 Not found) Is my syntax wrong? However this works: Isn't it possible to combine the two extension name="2020" into one extension 2020 Henk -- View this message in context: http://www.nabble.com/GotoIf-tp19793170p19794894.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] GotoIf
Brian West-3 wrote: > > Might want to try this: > > break="never"/> > When I try this I get a strange (looping ?) effect and not even the fist condition is executed (calling from 2000) > break="never"/> > > > > > > > Maybe it's not possible to test multiple caller_id_numbers inside one At least when I separate them in multiple extensions, it works. -- View this message in context: http://www.nabble.com/GotoIf-tp19793170p19803514.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] GotoIf
I discovered some small errors in previous threads concerning this item.
(transfer i.s.o bridge)
To conclude:
Asterisk:
---
exten => s,n,GotoIf($[${CALLERID(num)}=32476478861]?default,1000,1)
FS:
Henk
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[Freeswitch-users] Routing problem
I'm using http://www.voxbone.com for incoming virtual PSTN calls to
[EMAIL PROTECTED]
oegema.com is my (dyndns)domain and
voxbone is the extension in my domain (my FS server)
Based on the callerid number to route incoming PSTN calls, I've been using
following in Asterisk:
[default]
exten => voxbone,1,Goto(incoming,s,1)
[incoming]
exten => s,1,NoOP(Time=${STRFTIME(${EPOCH},,%H)}.${STRFTIME(${EPOCH},,%M)})
exten => s,n,GotoIf($[${CALLERID(num)}=32476378xxx]?default,2004,1)
exten => s,n,GotoIf($[${CALLERID(num)}=32486632xxx]?default,2005,1)
exten => s,n,GotoIf($[${CALLERID(num)}=0476378xxx]?default,2000,1)
.
.
exten => s,n,Goto(default,1000,1)
So far so good.
===
Now I want to do the same in FS.
If I don't have an extension "voxbone" and only use the following two
extensios:
then calls from 32476378xxx and from 32486632xxx are bridged correctly.
All other PSTN numbers are not routed anywhere. Which is correct.(because
"voxbone" don't exist)
However, when I add extension "voxbone"
then ALL calls are routed to 2000.([EMAIL PROTECTED] exist)
So the extensions and
are ignored.
It is however not possible to put multiple condition fields for
"caller_id_number" inside the "voxbone" extension
and
(like in Asterisk [incoming] exten =>s, )
See also discussion thread: http://www.nabble.com/GotoIf-td19793170.html
So I can't route my calls based on caller-id anymore.
So I'm still faced with the problem how to a route calls based on caller-id,
and if the called-id doesn't match, the calls should be routed to extension
"voxbone".
My apologies if the story is too long, but I didn't find another wy to
explain.
Henk
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Re: [Freeswitch-users] GotoIf
Brian West-3 wrote: > > You can bridge or transfer. If you wanted the exact behavior of > gotoif then transfer was it. Since you wanted to send it to extension > 1000 in context default. Using bridge doesn't do that. If you notice > in the default config their are things like voicemail, and various > other things that get set when you call 1000 but when you bridge you > bypass all that and the extension now has NO voicemail or any of the > features setup by calling 1000 directly in the dialplan. > > This doesn't work: But this does: I've been testing endless with these examples. That's why I came to the (wrong) conclusion that transfer doesn't work and bridge does. Until I saw my mistake: should have been Sorry for the confusion. You were so right Brian.:clap: -- View this message in context: http://www.nabble.com/GotoIf-tp19793170p19817097.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] GotoIf
Anthony Minessale-2 wrote: > > we also have the cond fsapi > > Thanks for the hint. Never knew about this one. -- View this message in context: http://www.nabble.com/GotoIf-tp19793170p19817205.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] GotoIf
Brian West-3 wrote: > > Now if you really wanna get fancy and you're running SVN trunk you can > do this: > > > > > > > > > Now if loopback is used and you come to a point where the underlying > channels bridge to each other the loopback channel will bow out and > leave the two channels bridged. > [EMAIL PROTECTED]> version FreeSWITCH Version 1.0.trunk (9841) [EMAIL PROTECTED]> 2008-10-05 00:16:26 [NOTICE] switch_channel.c:552 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [16089410-9262-11dd-ae2c-771a315e7f60] 2008-10-05 00:16:26 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing 2000->2020 in context default 2008-10-05 00:16:26 [ERR] switch_core_session.c:249 switch_core_session_outgoing_channel() Could not locate channel type loopback 2008-10-05 00:16:26 [ERR] switch_ivr_originate.c:964 switch_ivr_originate() Cannot create outgoing channel of type [loopback] cause: [CHAN_NOT_IMPLEMENTED] 2008-10-05 00:16:26 [INFO] mod_dptools.c:1848 audio_bridge_function() Originate Failed. Cause: CHAN_NOT_IMPLEMENTED -- View this message in context: http://www.nabble.com/GotoIf-tp19793170p19818399.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] How to get DISA working ?
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[Freeswitch-users] How to get DISA working ?
I'm trying to get DISA working. I've done this: The file disa.js is here: http://wiki.freeswitch.org/wiki/Examples_disa.js calling gives me busy tone: [EMAIL PROTECTED]> 2008-10-16 15:54:29 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [f3d6e460-9b89-11dd-910f-55dc10d13151] 2008-10-16 15:54:29 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing 2000-> in context default 2008-10-16 15:54:29 [NOTICE] switch_ivr.c:1116 switch_ivr_session_transfer() Transfer sofia/internal/[EMAIL PROTECTED] to [EMAIL PROTECTED] 2008-10-16 15:54:31 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-10-16 15:54:31 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/[EMAIL PROTECTED] [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-10-16 15:54:31 [NOTICE] switch_core_session.c:833 switch_core_session_thread() Session 12 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-10-16 15:54:31 [NOTICE] switch_core_session.c:835 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] And API call gives: [EMAIL PROTECTED]> jsrun /usr/local/freeswitch/conf/disa/disa.js 2008-10-16 15:56:42 [ERR] disa.js:27 mod_spidermonkey() TypeError: session.ready is not a function API CALL [jsrun(/usr/local/freeswitch/conf/disa/disa.js)] output: OK I'm I doing something wrong ? Henk -- View this message in context: http://www.nabble.com/How-to-get-DISA-working---tp20014227p20014227.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to get DISA working ?
Michael Jerris wrote: > > > Your routing to enum for extension and there is no enum route for > that number. > I should have seen that mistake myself. :blush: Does somebody have a script, similar to http://wiki.freeswitch.org/wiki/Examples_disa.js but without a password. (I don't have enough knowledge of java to program it myself) The way I'm using DISA is like this: 016123456pp So I call (from my mobile) 016123456, wait for a few seconds , and then send DTMF . (to dial extension ) Henk -- View this message in context: http://www.nabble.com/How-to-get-DISA-working---tp20014227p20038024.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] USER_NOT_REGISTERED (?)
I have one remote extension 2013 registered (?) to my FS server: Call-ID:NGQwNDQyMmY1ZTFjN2I1NDY4MGRlOGRiN2UzM2JhY2Q. User: [EMAIL PROTECTED] Contact:"Simon" Agent: Zoiper for Windows rev.599 Status: Registered(UDP)(unknown) EXP(2008-10-18 22:01:48) Host: ubuntu I can call FROM 2013 to all my local extensions, but not TO 2013 2008-10-18 22:16:54 [INFO] switch_ivr_async.c:1536 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-10-18 22:16:54 [ERR] switch_ivr_originate.c:1011 switch_ivr_originate() Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2008-10-18 22:16:54 [ERR] switch_ivr_originate.c:1011 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2008-10-18 22:16:54 [INFO] mod_dptools.c:1839 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED What must I do to call extensions in the domain 'oegema.com' ? Thanks Henk -- View this message in context: http://www.nabble.com/USER_NOT_REGISTERED-%28-%29-tp20050486p20050486.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] USER_NOT_REGISTERED (?)
henkoegema wrote: > > I have one remote extension 2013 registered (?) to my FS server: > > Call-ID: NGQwNDQyMmY1ZTFjN2I1NDY4MGRlOGRiN2UzM2JhY2Q. > User: [EMAIL PROTECTED] > Contact: "Simon" > > Agent:Zoiper for Windows rev.599 > Status: Registered(UDP)(unknown) EXP(2008-10-18 22:01:48) > Host: ubuntu > > I can call FROM 2013 to all my local extensions, but not TO 2013 > > 2008-10-18 22:16:54 [INFO] switch_ivr_async.c:1536 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf > XML features > 2008-10-18 22:16:54 [ERR] switch_ivr_originate.c:1011 > switch_ivr_originate() Cannot create outgoing channel of type [error] > cause: [USER_NOT_REGISTERED] > 2008-10-18 22:16:54 [ERR] switch_ivr_originate.c:1011 > switch_ivr_originate() Cannot create outgoing channel of type [user] > cause: [USER_NOT_REGISTERED] > 2008-10-18 22:16:54 [INFO] mod_dptools.c:1839 audio_bridge_function() > Originate Failed. Cause: USER_NOT_REGISTERED > > > What must I do to call extensions in the domain 'oegema.com' ? > > Thanks > Henk > I'm sure somebody knows the anwser to my problem. I really can't find it myself. :confused: -- View this message in context: http://www.nabble.com/USER_NOT_REGISTERED-%28-%29-tp20050486p20090027.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] USER_NOT_REGISTERED (?)
Brian West-3 wrote: > > double check your firewall. > I'm using an Edimax Wireless router with the Firewall module function Disabled. -- View this message in context: http://www.nabble.com/USER_NOT_REGISTERED-%28-%29-tp20050486p20095923.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] How to find an extension in public.xml ?
I have an offsite phone in ( as described in http://wiki.freeswitch.org/wiki/Example_Offsite_phones): //freeswitch/conf/dialplan/public.xml: = When I dial 2014, FS looks through the whole file in ../dialplan/default.xml, but can't find it there (because it's not definded in default.xml) (http://pastebin.freeswitch.org/6062) Q: How can I force FS to look in ../dialplan/public.xml if it can't find an entry in default.xml ? Henk -- View this message in context: http://www.nabble.com/How-to-find-an-extension-in-public.xml---tp20418918p20418918.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to find an extension in public.xml ?
Brian West-3 wrote: > > You could transfer to public at the end of default.xml > > /b > > How do I do that ? > ___ > 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 > > -- View this message in context: http://www.nabble.com/How-to-find-an-extension-in-public.xml---tp20418918p20422100.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to find an extension in public.xml ?
Brian West-3 wrote: > > You would use the transfer app. > > /b > Just something like: (??) at the end of the file default.xml ? -- View this message in context: http://www.nabble.com/How-to-find-an-extension-in-public.xml---tp20418918p20422829.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to find an extension in public.xml ?
Brian West-3 wrote: > > > > As per the wiki page > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer > This helps. :clap: -- View this message in context: http://www.nabble.com/How-to-find-an-extension-in-public.xml---tp20418918p20424174.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Linksys/SPA3000 port problem.
The Linksys/SPA3000 has two ports (1 FXS and 1 FXO port) The FXS interface (ext. 2014 in my case) is on port 5090 (profile name: doublenat and sip-port 5090) and 2014 is registered ok with FS. What to do with the FXO interface? This is what I've tried: I've created a new profile ( doublenat2) with sip-port 5091 ? I've put the FXO interface (ext. 2015) on port 5091, and 2015 is registerd ok with FS. Is this the right way froward?:working: I can call 2014 with: When I try to call 2015 with: the softphone shows: Call rejected: 480 temporarily Unavailble CLI: 2008-11-11 00:17:29 [ERR] switch_ivr_originate.c:1064 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2008-11-11 00:17:29 [INFO] mod_dptools.c:1848 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED The PSTN line in not actually connected o the FXO line. (I'm testing for somebody else. I don't have a PSTN line myself) Could that be the reason that it's not working properly? -- View this message in context: http://www.nabble.com/Linksys-SPA3000-port-problem.-tp20430421p20430421.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Strange (?) port number.
All my remote extensions are showing port number 5090, except the extension 2013. Why does it show port 56588 ? Call-ID:OTJlMGM3NWY4OTAyMDIwODEwODMwYjM5N2UyYjBiZGU. User: [EMAIL PROTECTED] Contact:"Simon" Agent: Zoiper for Windows rev.599 Status: Registered(UDP)(unknown) EXP(2008-11-13 11:59:51) Host: freeswitch Henk. -- View this message in context: http://www.nabble.com/Strange-%28-%29-port-number.-tp20478756p20478756.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Re comended VOIP Providers?
I'm using several of the Betamax (voipbuster, intervoip and more) providers and they all work well. Andy Ayers-2 wrote: > > Hi, > > Can anyone recommend any good VOIP providers that integrate well with > Freeswitch. In particular I need one that can cope with Freeswitch being > behind a firewall/router. I've tried Voiptalk.org and voipon.co.uk but > neither seem to register correctly via the router. > > Any help much appreciated. > > regards > Andy > > > ___ > 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 > > -- View this message in context: http://www.nabble.com/Recomended-VOIP-Providers--tp20462805p20478821.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Strange (?) port number.
Brian West-3 wrote: > > Because thats the contact's port number. > But I programmed port 5090. -- View this message in context: http://www.nabble.com/Strange-%28-%29-port-number.-tp20478756p20479337.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Nested conditions not allowed.
How do I program following: I call extension 1234. Only if the time is between 9 am and 6 pm then play the file were-sorry.wav I tried: but that doesn't work, because nested conditions are not allowed. How do I solve a problem like this. -- View this message in context: http://www.nabble.com/Nested-conditions-not-allowed.-tp20624343p20624343.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Nested conditions not allowed.
Where in the wiki ? Anthony Minessale-2 wrote: > > instead you stack the conditions and they must all be true to proceed to > the > next unless you change the break attribute (see wiki) > > > On Fri, Nov 21, 2008 at 10:14 AM, henkoegema > <[EMAIL PROTECTED]>wrote: > >> >> How do I program following: >> >> I call extension 1234. >> Only if the time is between 9 am and 6 pm then play the file >> were-sorry.wav >> >> >> I tried: >> >> >> >> > expression="^((09|1[0-7])[0-5][0-9]|1800)$"> >> > data=".../.../sounds/were-sorry.wav"/> >> >> >> >> >> but that doesn't work, because nested conditions are not allowed. >> >> How do I solve a problem like this. >> -- >> View this message in context: >> http://www.nabble.com/Nested-conditions-not-allowed.-tp20624343p20624343.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> ___ >> 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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > iax:[EMAIL PROTECTED]/888 > googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > pstn:213-799-1400 > > ___ > 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 > > -- View this message in context: http://www.nabble.com/Nested-conditions-not-allowed.-tp20624343p20626999.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Nested conditions not allowed.
Brian West-3 wrote: > > Also look at the global extension in the default config.. that is an > example of stacking them. > Tnx. =) -- View this message in context: http://www.nabble.com/Nested-conditions-not-allowed.-tp20624343p20630498.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] strftime(%u)
strftime(%u) never gives me any output in an application ! While strftime(%w) does. (output 6 (saterday) ) Is this a bug or could there be any other reason? Just as a test, I tried: output: _to_2000_from_2003.wav - output: 6_to_2000_from_2003.wav -- View this message in context: http://www.nabble.com/strftime%28-u%29-tp20635116p20635116.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] strftime(%u)
Brian West-3 wrote: > > what platform are you on? > > On Nov 22, 2008, at 4:44 AM, henkoegema wrote: > >> >> strftime(%u) never gives me any output in an application ! While >> strftime(%w) does. (output 6 (saterday) ) > Linux: Ubuntu 8.04 ( 2.6.24-19-server) FS: Version 1.0.trunk (10272) -- View this message in context: http://www.nabble.com/strftime%28-u%29-tp20635116p20639213.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Transfer/Bridge
What is the difference between and ? -- View this message in context: http://www.nabble.com/Transfer-Bridge-tp20647162p20647162.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Transfer/Bridge
Iñaki Baz Castillo wrote:
>
> El Domingo, 23 de Noviembre de 2008, henkoegema escribió:
>> What is the difference between
>>
>>
>> and
>> ?
>
> AFAIK "transfer" means jumping to other dialplan section ("1000" in your
> case)
> while "bridge" means generating a new call (leg B) and bridge it with leg
> A
> (the caller).
>
But what difference does that make when I call 1000 ?
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Re: [Freeswitch-users] Transfer/Bridge
Iñaki Baz Castillo wrote: > > El Domingo, 23 de Noviembre de 2008, henkoegema escribió: >> Iñaki Baz Castillo wrote: > > - "transfer" will jump to the dialplan "1000" extension, and that can do > anything you have decided in that dialplan. > > Isn't bridge also executing the dialplan "1000" extension ? -- View this message in context: http://www.nabble.com/Transfer-Bridge-tp20647162p20648836.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Channel variable 'call_timeout'.
I use following diaplan for extension 2005: - What I can't get to work is the "call_timeout" variable. It is set to value 10 at the moment. So I was thinking, that if exten. 2005 is busy or doesn't answer, the call would be transferred to the voicemail after 10 seconds. It is actually transferred after 66 (?) seconds. Changing the value of call_timeout doesn't seem to have any effect. :confused: -- View this message in context: http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20677406.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] I am a new user
Faisal Maqsoodi wrote: > > I ve just installed freeswitch on my system. What should i try on it now > to take a start? I ve gone through the theory of pbx and all that. faisal > http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/SOHO_PBX_Example -- View this message in context: http://www.nabble.com/I-am-a-new-user-tp20678693p20678807.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Channel variable 'call_timeout'.
Doens't make any difference. :working: Michael Jerris wrote: > > try using export instead of set for that var. > > Mike > > On Nov 25, 2008, at 4:05 AM, henkoegema wrote: > >> >> I use following diaplan for extension 2005: >> - >> >> >> >> > data="continue_on_fail=USER_BUSY,NO_ANSWER"/> >> >> >> >> >> >> What I can't get to work is the "call_timeout" variable. >> It is set to value 10 at the moment. >> >> So I was thinking, that if exten. 2005 is busy or doesn't answer, >> the >> call would be transferred >> to the voicemail after 10 seconds. >> >> It is actually transferred after 66 (?) seconds. >> >> Changing the value of call_timeout doesn't seem to have any effect. >> :confused: > > > ___ > 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 > > -- View this message in context: http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20682038.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Channel variable 'call_timeout'.
Anthony Minessale-2 wrote:
>
> also try
>
> data="{originate_tiimeout=10}sofia/internal/2005%$${domain}"/>
>
YES, that works !!
But... why doesn't
work ?
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Re: [Freeswitch-users] Channel variable 'call_timeout'.
Michael Collins-11 wrote: > > On Tue, Nov 25, 2008 at 11:44 AM, Anthony Minessale < > [EMAIL PROTECTED]> wrote: > >> iirc call_timeout has been changed to originate_timeout >> > > That seems to jive with what's in switch_ivr_originate.c as there is no > mention of "call_timeout" anywhere in the function switch_ivr_originate(), > but "originate_timeout" is definitely there. > > Question: I see that "call_timeout" is in function > switch_ivr_wait_for_answer() but "originate_timeout" is not, and it > definitely looks like the call_timeout channel variable is being used to > populate the integer "timeout". (See switch_ivr_originate.c:332) > > I was wondering if "call_timeout" was the correct channel variable here or > should it be "originate_timeout"? Just confirming. > > Thanks, > MC > > I used "call-timeout" because that's how I understood it from the Wiki. (?) -- View this message in context: http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20689832.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] How to assign a value to a variable from system call.
How do I do this in FS ?
exten =>
3000,n,Set(status=${CURL(http://localhost/asterisk/credit/credit_dialnow.php)})
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Re: [Freeswitch-users] How to assign a value to a variable from system call.
Anthony Minessale-2 wrote: > > 1) Open your browser to the bounty page > 2) Post a bounty to add that feature. > Do you mean this feature is not available in FS ? -- View this message in context: http://www.nabble.com/How-to-assign-a--value-to-a-variable-from-system-call.-tp20700309p20706196.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] How to assign a value to a variable from system call.
Anthony Minessale-2 wrote: > > 1) Open your browser to the bounty page > 2) Post a bounty to add that feature. > What (where) is the bounty page ? -- View this message in context: http://www.nabble.com/How-to-assign-a--value-to-a-variable-from-system-call.-tp20700309p20707554.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
Re: [Freeswitch-users] Failed to create a udp socket on port 5060
Faisal Maqsoodi wrote: > > Yes, i m running both on the same machine. Now i changed it to port 5020 > for twinkle. But the error msg displayed is: > Sat 10:37:25 > 1001, registration failed: 503 Service Unavailable > What should i try? > Make a new profile with port 5020. -- View this message in context: http://www.nabble.com/Failed-to-create-a-udp-socket-on-port-5060-tp20732432p20758094.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] libtool version.
r...@msi:/home/henkoegema/freeswitch# ./bootstrap.sh bootstrap: checking installation... bootstrap: autoconf version 2.61 (ok) bootstrap: automake version 1.10.1 (ok) bootstrap: libtool version 2.2.4 found. You need libtool version 1.5.14 or newer installed <--Isn't that what I have :confused: to build FreeSWITCH from SVN. r...@msi:/home/henkoegema/freeswitch# -- View this message in context: http://www.nabble.com/libtool-version.-tp2127p2127.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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
