September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SRTP and bridge woes
by Daniel Ivanov 25 Sep '12

25 Sep '12

I am building an ostn node. I followed a recipe, but got caught up with learning FS basics. I had this trouble that calls were being signalled, but no audio was actually put through. I thought it was nat problems, but connectivity was going fine. This is my dialplan snippet: <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>  <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>  <action application="info"/> <action application="answer"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/>  <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> This works, but you can see i have an answer app right before the bridge. If i don't answer the call, i don't get to the point where i get Activating SRTP RECV/SRTP SEND. Yet both channels get answered and RTP goes both ways, but i have a feeling it's not SRTP and thus is not being decrypted. I have this in the global context: <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/>  <action application="export" data="sip_secure_media=true"/> </condition> which should activate SRTP on both legs. I have pasted in to the bin a debug log with and without the answer and media sending here : http://pastebin.freeswitch.org/19937

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Consultation Call via event_socket interface
by Alexander Haugg 25 Sep '12

25 Sep '12

Hi All, I'm new on the mailing list. I have a problem with a call scenario. - Channel A and channel B are bridged (A is my own channel and B is my calling partner) - Now i set channel B on hold with the command "uuid_hold xxx" and create a new channel to C with the command: bgapi originate {channel_csid=num,accountcode=num,origination_caller_id_name='num@ip',origination_caller_id_number=num@ip}sofia/external/num@ip &park this works correctly, the partner C answer and the channel is established. - Now the Problem: I try to bridge the channel a to channel c with the uuid_bridge command, now the channel b will hangup, why? The variable hangup_after_bridge is by default false. Other problem: channel A can hear the voice of channel C but not speak with him, channel C can hear and speak. But this problem is not the important think at the moment. Is there a general problem in my plan to do that? Is it a better plan to do this over the dialplan? The next step in this scenario is to toggle the connection A -> B and A -> C. Thanks for your help! Nice regards, Alex

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Apache
by onyeagbaikenna04＠gmail.com 25 Sep '12

25 Sep '12

Hi everyone How can I get freeswitch running on apache 2.. Thanks for your anticipated response Sent from my BlackBerry smartphone from Virgin Media

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Mod curl, user directory and Gtalk
by Peter van Raamsdonk 25 Sep '12

25 Sep '12

Morning list, I posted my question before but want to elaborate a bit... For the mod curl my web servlet returns the user directory extensions in xml, <document type="freeswitch/xml"> <section name="directory" description="user directory"> <domain name="my.domain">  <user id="1001"> <params> <param name="password" value="xxxx"/> <param name ="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1001"/> <variable name="effective_caller_id_number" value="1001"/> <variable name="user_context" value="default"/> </variables> </user> </domain> </section> </document> This works fine and the SIP phones are found by FS. Is it possible to include also Gtalk profile params in the extension to register Gtalk extensions? http://wiki.freeswitch.org/wiki/Dingaling#Sample_Configuration_.28Google.29 Kind regards, Peter

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Freeswitch on Apache
by onyeagbaikenna04＠gmail.com 24 Sep '12

24 Sep '12

Hi everyone How can I get freeswitch working on apache 2.. Thanks for your anticipated response Sent from my BlackBerry smartphone from Virgin Media

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Timezone with time conditions
by Spencer Thomason 24 Sep '12

24 Sep '12

Hello all, Has anyone found an elegant solution to using time conditions with multiple locations? For example suppose you have a multi tenant system with time conditions, is there any way to set the call time zone on a per call basis? Thanks, Spencer

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Newbie question, FS cli and db (users for user directory)
by Peter van Raamsdonk 24 Sep '12

24 Sep '12

Hi all, Just wondering if I can add/ delete users to the db from the CLI (I'm using ESL java library)? See http://wiki.freeswitch.org/wiki/Function_db >From the Dialplan this is possible, I want however to dynamically add new users. <action application="db" data="insert/realm/key/value"/> In the api (mod commands) nothing is mentioned about "db" I don't know if something like this could work with my example extension 1999, default group named default... response = client.sendSyncApiCommand( "db", "insert/default/1999/1999"); Another option is to use mod_xml_curl and feed the dialplan or user directory to FreeSwitch back again. Kind regards, Peter

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Fifo agent status and callerid
by Peter van Raamsdonk 24 Sep '12

24 Sep '12

Hi all, I'm implementing fifo. I have registered my agents succesfully and drop a callerid in my queue. The phones (agents) start to ring and show the correct callerid. However the status I receive is from the caller and I expected this would be from the agents (the agent is being called and not the consumer). The caller-callerid-number should be of the agent I believe? Should I add something like fifo_orbit_exten or fifo_consumer_id ? Regards Peter

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Git checkout failed
by Виталий Давудов 24 Sep '12

24 Sep '12

Hi, list! I try to update my freeswitch to stable version 1.2. According to FS Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Selecting_version) I applied this command in previos crated freeswitch.git directory: git checkout v1.2.stable But system returned me this error: error: pathspec 'v1.2.stable' did not match any file(s) known to git. What did I do wrong? -- Best regards, Vitaly.

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FreeSWITCH Weekly News and Notes
by Michael Collins 24 Sep '12

24 Sep '12

Welcome to the last Monday of September 2012! We've had quite the interesting week. Perhaps the most interesting item the team dealt with was a vulnerability in the Sofia SIP stack that would cause a segmentation fault while processing a specially crafted SIP message. Just to show you how nimble the FreeSWITCH developers are, from the time the vulnerability was reported it took less than a day to fix, test, and roll a new version of FreeSWITCH. We encourage everyone on 1.2 to get updated to version 1.2.3 <http://files.freeswitch.org/freeswitch-1.2.3.tar.bz2> as soon as possible. (The fix is also in the 1.3 development branch as of last Wednesday, September 19.) We tip our hats to Anthony and the rest of the dev team<http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_So_where_do_I_find_wishli…>for their hard work on our behalf. Last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19>was also very informative. We received an introduction to the repro SIP proxy <http://resiprocate.org/About_Repro> software. We look forward to this coming Wednesday <http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26>where Scott Godin and Daniel Pocock will continue the discussion and will get deeper into how to set up the proxy and use it with FreeSWITCH. If you haven't already tried to install repro please do so. Daniel has a nice tutorial<http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch>over at OpenTelecoms.org - be sure to check it out and bring your questions on Wednesday. Finally, we'd like to draw your attention to this blog post<http://blog.krisk.org/2012/09/freeswitch-stable-branch-sponsored-by.html>by long time FreeSWITCH and open source telephony supporter Kristian Kielhofner. Kristian reports that his company, Star2Star Communications<http://www.star2star.com/>, is sponsoring the FreeSWITCH stable branch by giving direct financial support to the project. This allows for a full-time team member to work on things like the stable branch and packaging as well as community interaction and documentation. We appreciate those who support FreeSWITCH and open source telephony! Have a good week and we'll see you again in October. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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