April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Originate call into conference - no early media
by Alex Crow 23 Apr '12

23 Apr '12

Hi, I'm using the following dialplan snippet to call new parties into a conference (see : http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example) <condition field="destination_number" expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true"> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/> <action application="hash" data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" /> <action application="set" data="res=${bgapi originate {origination_uuid=${new_uuid}}user/$1 &conference(${conference_name})}"/> </condition> <condition field="destination_number" expression="^ADD_CALL_TO_CONF__(2\d{2})$" break="on-true"> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/> <action application="hash" data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" /> <action application="set" data="res=${bgapi originate {origination_uuid=${new_uuid}}sofia/gateway/om.net/$1 &conference(${conference_name})}"/> </condition> The page referenced implies I should get early media when I call someone into the conference, but when I try (either for local endpoints or those on a remote server via a gateway) I don't hear any ringing. I do if I call via a PSTN gateway. Any ideas? Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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FreeSWITCH News and Notes
by Michael Collins 23 Apr '12

23 Apr '12

Greetings everyone. We have several items for your consideration this week: - The FreeSWITCH Community Conference Call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_25>will take place at 1PM Eastern, 10AM Pacific this coming Wednesday April 25. We will be having an informal discussion about executing dialplan applications or APIs using the "execute_on" and "api_on" family of channel variables. We look forward to having you there with us. - The authors of the FreeSWITCH Cookbook will be taking part in an interview on the weekly VUC call <http://www.voipusersconference.org/>this Friday April 27. - Vestec still has some ASR starter kits to give away to FreeSWITCH users around the world. If you have need of speech recognition in your telephony application then contact marketing(a)cluecon.com and ask for more information. Vestec already supports languages in 10 of the largest economies of the world and more languages are in development. Please make plans to support both conference calls this week. If you've never been to either one I highly recommend checking them out. They are very interesting. In fact, they are a small taste of what ClueCon is like: hanging out with fellow telecom enthusiasts and having technical discussions. It really is a lot of fun. Speaking of ClueCon, be sure to register <http://www.cluecon.com/register/> today. Don't forget that if you register in April you get extra entries in the giveaway drawings we have throughout the conference. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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unimrcp nuance grammar failed to load
by Anita Hall 23 Apr '12

23 Apr '12

Hi I am getting the following error when using detect_speech from dialplan with unimrcp, nuance speech server and nuance recognizer. I understand that it is the job of the unimrcp client (ie mod_unimrcp) to convert the grammar from JSGF to MRCP. Why do I not see any grammar text in the SIP trace ? What could be wrong here ? 2012-04-21 01:58:01.855558 [DEBUG] mod_unimrcp.c:3433 (ASR-0) grammar failed to load, status code = 407 2012-04-21 01:58:01.855558 [DEBUG] mod_unimrcp.c:1541 (ASR-0) PROCESSING ==> ERROR Complete logs are here http://pastebin.freeswitch.org/18900 The SIP trace : (10.60.20.25 is FreeSWITCH and 10.60.20.47 is Nuance Speech Server) 10.60.20.25 -> 10.60.20.47 SIP/SDP Request: INVITE sip:10.60.20.47:5060, with session description 10.60.20.47 -> 10.60.20.25 SIP/SDP Status: 200 OK, with session description 10.60.20.47 -> 10.60.20.25 SIP Request: BYE sip:10.60.20.25:5090 10.60.20.25 -> 10.60.20.47 SIP Status: 481 Call Does Not Exist My dialplan is <action application="answer"/> <action application="detect_speech" data="unimrcp:nuance5-mrcp2 yesno yesno.gram"/> <action application="detect_speech" data="grammar yesno"/> <action application="detect_speech" data="grammaron yesno"/> <action application="sleep" data="500"/> My grammar is yesno.gram kept in /usr/local/freeswitch/grammar #JSGF V1.0; grammar example; <yes> = [ yes ]; <no> = [ no ]; public <results> = [ <yes> | <no> ]; # cat conf/autoload_configs/unimrcp.conf.xml is here http://pastebin.freeswitch.org/18899 # cat conf/mrcp_profiles/nuance-5.0-mrcp-v2.xml is here http://pastebin.freeswitch.org/18898 regards, Anita

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Mixing dynamic (xml_curl) and static (XML) dialplans
by Weigel, Stefan 23 Apr '12

23 Apr '12

Hi, I'm trying to mix static (XML file) and dynamic (mod_xml_curl) dialplan content. My static dialplan looks like: /tmp/test.xml: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="public"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <extension name="outside_call" continue="true"> <condition> <action application="set" data="outside_call=true"/> </condition> </extension> <extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> </condition> </extension> </context> </section> </document> I changed parameter "dialplan" in profile internal.xml: <param name="dialplan" value="XML:/tmp/test.xml,XML"/> But when placing a call I only see that diaplan from static XML is parsed and executed (message "has executed the last dialplan instruction, hanging up."). What's going wrong ? Thanks and cheers Stefan

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(no subject)
by Brian Foster 22 Apr '12

22 Apr '12

Hola, When I recieve a fax, I do something like this: <extension name="fax"> <condition field="destination_number" expression="fax"> <action application="answer"/> <action application="sleep" data="1"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="set" data="execute_on_fax_success=system /usr/local/freeswitch/scripts/success_postfax.sh '/tmp/FAX-${uuid}.tif' '${fax_result_code} '${fax_result_text}' "/> <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/> <action application="hangup"/> </condition> </extension> I'm actually getting a few other fax-related variables, but I figured I might shorten it up a bit so it wasn't extremely long on the email. Anyway, the fax variables I'm trying to pull aren't showing up when I try and do my post-fax processing when receiving the fax. Am I doing this all wrong? Should I be doing something to keep the variables set by mod_spandsp alive until I do my postfax routine? -BDF -- Brian D. Foster Endigo Computer LLC Email: bdfoster(a)endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.

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Group_confirm_key on one channel only (using square brackets)
by Brett Wilson 21 Apr '12

21 Apr '12

Hey guys, I am trying to do a simulataneous bridge to an internal extension as well as a cell phone. Of course there is the issue of the phone's voicemail picking up if the call goes too long or if the phone is off, or the call is rejected. Group confirm key works well for this situation however I cannot seem to get it to work with square braces, only curly ones. I don't want confirm key to work on the internal extension, just the cell phone. Here is a portion my dialplan: However it does not work like this, only with curly braces. Any idea how I can accomplish this? -- ******************************************* BRETT WILSON IT DEPARTMENT LAUNCH 3 VENTURES, LLC 134 Myer Street Hackensack, NJ 07601 PHONE: 877.878.9134 FAX: 646.536.3866 EMAIL: Brett.Wilson(a)launch3.net [1] AOL IM: Brett.Wilson(a)launch3.net [2] www.Launch3.net [3] www.Launch3telecom.com [4] ******************************************* [5] [6] [7] [8] Links: ------ [1] mailto:Brett.Wilson@launch3.net [2] mailto:Brett.Wilson@launch3.net [3] http://www.launch3.net/ [4] http://www.launch3telecom.com/ [5] http://launch3telecom.blogspot.com/ [6] http://www.facebook.com/pages/Launch-3-Telecom/165610820178720 [7] https://twitter.com/launch3telecom [8] https://plus.google.com/102713212642017290959

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mod_gsmopen, now with USB dongle support (SMSs and Voice calls to/from Cellular Networks)
by Giovanni Maruzzelli 21 Apr '12

21 Apr '12

Hello FreeSWITCHers, the new version of GSMopen is basically usable, and the docs at the wiki almost up to date (barring the details of the config file, you don't need them now :) ). Good news is that GSMopen now supports Huawei E1550 and compatible dongles, that are cheap (roughly $20 on Ebay) and reliable. Thanks to dongle's pcm serial streaming via USB, no more need for sound hacks, ALSA, or any sound subsystem (and cpu load). All is digital. It's reasonable to expect to be able to support many dozen concurrent calls per machine (dongles are USB 2.0 compliant). Works well right now on Linux and Windows, and would probably works on *BSD and OSX with minor tweaks. Code will be polished, refactured, rewritten, etc. But the thing is there. So, please update your source code to latest git, have a look at the wiki page, build mod_gsmopen, connect a dongle to your machine, and begin catching bugs and documentation inaccuracies. And asking for features for your use cases. Thanks to you all in advance, it's a pleasure and an honor to be part of this community. http://wiki.freeswitch.org/wiki/Gsmopen -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Dropped calls
by Rob Morin 21 Apr '12

21 Apr '12

I recently (4/17/2012) did a "make current" on my Freeswitch PBX. (Last time I had done it was probably in January.) No particular reason for the update - just timely. After the update, we're now seeing calls drop consistently at the ~1:47 mark. Obviously, this wasn't the case beforehand. I can provide a pcap file and Freeswitch log of a call demonstrating the issue; I'm just not sure what the best method to do so is. The FS server is sending a BYE, the REASON is 'SIP; cause=408; text="Session timeout"'. On my end, I have a Grandstream 502 and I can see the keep-alives are working. But now it is getting disconnected while it wasn't before the update. Any ideas? Thank you, Rob

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FreeSWITCH mod_fail2ban
by Kyle King 21 Apr '12

21 Apr '12

Hello everyone, I've written a FreeSWITCH module for the application Fail2ban. It currently logs all inbound registration attempts, and failed registrations to a log file "/usr/local/freeswitch/log/fail2ban.log". The advantage to using this module for fail2ban rather than the standard FS log is that you don't have to keep your server log level as verbose. If anyone is willing to look it over and give me feedback please do so at https://github.com/kyconquers/freeswitch-mod_fail2ban -- Kyle A. King Quentus Technologies, INC Cell: 703-635-9512 Work: 253-218-6030 Fax: 206-462-1861 Email: kyle.king(a)quentustech.com

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Re: [Freeswitch-users] Possible registration/presence bug - Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension registration is present
by Phil Quesinberry 21 Apr '12

21 Apr '12

FYI, this is still a problem as of FreeSWITCH Version 1.1.beta1 (git-735e96a 2012-04-19 19-46-11 -0700). I can generally get it to occur if I do a full reboot of the XCP Dom0 hypervisor it's running on. As before, restarting FS fixes the problem. Everything looked ok according to 'sofia status profile internal reg'. I don't have time to figure out how to file a Jira tonight but I'll see if I can get one filed this weekend... we're going to Hershey Park tomorrow. Cheers, - Phil ---------- Anthony Minessale anthony.minessale at gmail.com <mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users% 5D%20Possible%20registration/presence%20bug%20-%0A%20Sometimes%20get%20%22Or iginate%20Failed.%20Cause%3A%20USER_NOT_REGISTERED%22%20when%20extension%0A% 20registration%20is%20present&In-Reply-To=4F7A0B04.2070405%40integrafin.co.u k> Tue Apr 3 01:02:06 MSD 2012 * Previous message: [Freeswitch-users] Possible registration/presence bug - Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension registration is present <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/082516.ht ml> * Next message: [Freeswitch-users] Possible registration/presence bug - Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension registration is present <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/082523.ht ml> * Messages sorted by: [ date ] <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/date.html > [ thread ] <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/thread.ht ml> [ subject ] <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/subject.h tml> [ author ] <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/author.ht ml> I tracked this down to a race on startup where the core has not loaded the ACL lists but sofia tries to use them for NAT detection. I'm sure this is not something new but it should be fixed in HEAD. (I WISH SOMEONE WOULD HAVE OPENED A JIRA TO REFERENCE) _____________________________________________ From: Phil Quesinberry [mailto:philq@qsystemsengineering.com] Sent: Monday, April 02, 2012 3:44 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: Possible registration/presence bug - Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension registration is present The phone never stops working. The problem is either present from the time FS is started or it isn't. I'll have to bring it down and try to get it to fail again after hours so I can provide more detailed info. To me, it appears from the log I captured that presence_id is not always being initialized properly when FS is started but I haven't looked at the source code to see what's really happening. I do have the two FS books on the way though, which should give me a deeper understanding of how things work. The wiki only goes so far. Also, when I do a sofia status profile internal reg, I see a bunch of long expired registration entries. Not sure if that could somehow be related to this problem.

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