May 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Playing with ACL and authenticate users
by Elhodred 15 Jan '15

15 Jan '15

Hi all, I was playing around trying to have some users authenticated by IP and some users to be challenged for username/password auth. The problem is that I don't get it all. If I configure 2 users in this way: <user id="6001" cidr="192.168.28.12/32"> <params> </params> <variables> <variable name="accountcode" value="6001"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> <user id="6002"> <params> <param name="password" value="111111"/> </params> <variables> <variable name="accountcode" value="6002"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> And I added in sofia.conf.xml <param name="apply-inbound-acl" value="domains"/> User 6001 can authenticate if it's using IP 192.168.28.12 and can place calls, but user 6002 is rejected by acl "domains" Then I tried this: <user id="6001"> <params> <param name="auth-acl" value="192.168.28.12/32"/> </params> <variables> <variable name="accountcode" value="6001"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> <user id="6002"> <params> <param name="password" value="111111"/> </params> <variables> <variable name="accountcode" value="6002"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> And I deleted in sofia.conf.xml <param name="apply-inbound-acl" value="domains"/> and added <param name="auth-calls" value="true"/> Now user 6002 can place calls but for user 6001 I see in the log : 2013-05-23 18:15:27.222090 [WARNING] switch_core_state_machine.c:514 bed463c1-742e-4e24-829f-a6188a6667b3 sofia/GOLDEN/XXXXXXXX(a)XXX.XXX.XXX.XXX Abandoned 2013-05-23 18:15:27.222090 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/GOLDEN/XXXXXXXX(a)XXX.XXX.XXX.XXX [CS_NEW] [WRONG_CALL_STATE] This is my dial plan config: <context name="GOLDEN"> <extension name="unloop"> <condition field="$${unroll_loops}" expression="^true$"/> <condition field="$${sip_looped_call}" expression="^true$"> <action application="deflect" data="$${destination_number}"/> </condition> </extension> <extension name="outside_call" continue="true"> <condition> <action application="set" data="outside_call=true"/> </condition> </extension> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="hangup"/> </condition> </extension>  <extension name="remove_prefix_2" continue="true"> <condition field="network_addr" expression="^XXX\.XXX\.XXX\.XXX$"/> <condition field="destination_number" expression="^12345(\d+)$"> <action application="log" data="Removing leading digits"/> <action application="set" data="destination_number=$1"/> </condition> </extension> <extension name="distributor_TEST2_37360______"> <condition field="${sip_authorized}" expression="^true$" break="never"/> <condition field="destination_number" expression="^(\d+)$"> <action application="log" data="Dialing User"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="user/$1(a)XXX.XXX.XXX.XXX"/> </condition> </extension> And now I'm lost. Please, can anybody point me where the issue can be? Do I need to have different sofia profiles, one for IP authentication and one for username/password auth? Regards, Alfonso.

6 7

Transfer hangup but b-leg ring
by FERNANDO VILLARROEL 29 Dec '14

29 Dec '14

Hi. I am trying to use nibblebill and lcr. If originator (example 1006) account have zero balance and trying to call to 1004, the call hangup but the phone 1004 ring anyway. My dialplan: <context name="default"> <extension name="Outbound to PSTN 11 Digits"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="lcr" data="$1 pg_prefix2"/> <action application="set" data="nibble_account=${accountcode}"/> <action application="set" data="nibble_rate=${lcr_rate_1}"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="hangup"/> </condition> </extension> The log of call: http://pastebin.freeswitch.org/19228 How i can do?

2 1

Billing software
by Luis Daniel Lucio Quiroz 09 Dec '14

09 Dec '14

What other options for FS compatible software other than vBilling do you recommend me?

7 12

Recommendation for Billing for Hosted PBX Platform
by Deon Vermeulen 07 Dec '14

07 Dec '14

Hi, I've been looking at the Kazoo as a very viable option as a HostedPBX platform, but don't find any information with regards to Billing. I've also been tracking the thread about recommended GUI where 2600Hz are in the process for updated the code in git for Blue.box. This also looks like a possibility for a HostedPBX platform, but then again I have no idea about the Billing software to use for this. FusionPBX is a top contender if no the number 1 for a HostedPBX platform but again information for Billing is lacking. I would like to know what the feel out there is for Billing on a HostedPBX Platform, i.e. Cloud. If would be great if there is an opensource Billing system. I've purchased the jBilling Telecom guide, but no where can I find information about integrating with FreeSWITCH nor implementing it for a HostedPBX platform in the Cloud. Kind Regards

5 4

what can I have in ${} construction
by covici＠ccs.covici.com 11 Nov '14

11 Nov '14

Hi. I was experimenting with ${variable} and was wondering what could be in there before the }? I can do a substring if I say variable:2, but I wonder what else you can do? The bash manual has all kinds of things you can have, but how much of this will fs do? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

5 5

XML DialPlan help
by Sean Devoy 04 Nov '14

04 Nov '14

I swear I have read this, but now that I need it I can't find it. My user wants to dial international. I want to require them to do SOME extra step other than dialing 0113010... to call the Netherlands. Oddly enough, it is cheaper than US termination - go figure. Ideally, I would like them dial some extension like 777, hear a new dialtone and then accept the international number. I might even like to play a message that says "All charges for International are their responsibility and not part of their current package" before the dialtone. Ideas? Thanks again, Sean

3 4

create dummy channel to record conference
by Alexandre Mouillé 07 Oct '14

07 Oct '14

Hi, I hope I'm not posting this in the wrong mailing list. I'm working on a custom conference that uses the conference module. I need a record function that keeps recording (or at list appends the same wav file) when no one is in a given conference. The solution that was used in the Asterisk program I'm porting was to create a dummy channel that goes in the conference and records what it hears in a file. - Can I create a dummy channel with an "originate /loopback" that would go into a conf and record what it hears (I'm not sure what loopback really is) - Otherwise, can I use originate from the FS box to call itself, one side connects to the conference and the other records. (I'm not sure how to do that, and how I could transmit variables (to tell which file to record to) to the called program) Thanks

5 8

Goip GSM Gateway works great with FreeSwitch!
by Dissident 12 Sep '14

12 Sep '14

http://freeswitch-users.2379917.n2.nabble.com/file/n6691087/single-channel-… Hi, I've configured this GSM gateway with Freeswitch and it works great. I'm posting this so that if somebody tries to configure this Gateway won't have to spend as much time as a I did :) This is the configuration: *Gateway* *freeswitch/conf/sip_profiles/external/goip.xml* *Tips to check that everything is OK - Remember to remove the PIN number from your SIM card. Double check your SIM card with a Phone. It should not ask you for a PIN. Disable it! Period :) - *sofia status profile internal* (the Goip extension user must appear, I chose ext. 1000 for it...) - * sofia status* (to check that the Goip gateway is properly configured. - On the Goip status page the Phone Status and your GSM Status must be LOGIN - On the Goip status page GSM operator GSM signal must show something. If it doesn't it's because it has not logged properly with your GSM Sim card. Double check it with a Phone. - If the Goip is logged properly with FS you can use your favorite SIP client to log directly into GOIP that is to say. if your GOIP IP is 192.168.2.2 you can log into 1000(a)192.168.2.2 *(again this is not your FS IP is the GOIP IP)* and make an outbound call to an ordinary cell phone. This is useful in case you have trouble with the outbound call. * Things to find out :?? - First I used X-Lite, Windows XP and miniSip server 20 clients to test this gateway. It's kind of strange that with miniSip every time I call to the extension I chose for Goip a tone appears an then I can dial a cell phone number. FS says that this extension is not available, it shows up with the sofia status profile internal though. *I've used with Linux Ubuntu: - FreeSWITCH Version 1.0.head (git-decfdbb 2011-08-11 14-15-26 -0500) - Jitsy Java Sip Client - 3 Android Phones with CSipSimple, Sipdroid and 3CXPhone Sip Client - 2 linksys spa941 - 1 Goip GSM GateWay http://goip.com.ua/download/goip-series-manual.pdf http://goip.com.ua/download/goip-series-manual.pdf http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&In… http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&In… I hope you find this info useful, if you have any comments or requests or corrections... please let me know... best regards. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Goip-GSM-Gateway-works-great-… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 5

Skype is likely to stop supporting desktop API by Dec. 2013
by Henry Huang 12 Sep '14

12 Sep '14

Here is the link: http://www.engadget.com/2013/11/02/skype-killing-old-api/?ncid=rss_truncated I don't know if this mean that we'll no longer be able to interconnect with Skype anymore. And I just want to share the news with this community. Henry

8 8

mod_curl setting content type header
by Jeff Stark 04 Sep '14

04 Sep '14

Hi, Is there a way to set the Content-Type header when using mod_curl in a Lua script? Using curl on the command line I can do the following: curl -H "Content-Type: application/json" -H "Accept: application/json" -X POST -d '{"field1":"val1","field2":"val2"}' http://mydomain.com/ And in a Lua script I can do the following: local my_json = "{\"field1\":\"val1\",\"field2\":\"val2\"}" api = freeswitch.API() response_data = api:execute("curl", "http://mydomain.com/ post" .. my_json) But I don't see a way in the freeswitch API to set the Content-Type header in the above script. Appreciate any help or insight that can be offered. Thanks. Jeff

3 4