September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Project Looking For Volunteers: Web Site Dev/Maint
by Michael Collins 13 Nov '12

13 Nov '12

Hello all, I need to refresh my list of people who have Web dev skills and are willing to assist with things like Web site design, programming, and maintenance. If you have any of the following skills, or know someone who does and who is willing to donate a few hours, please email me offlist: HTML5 CSS2/3 Javascript/JQuery Drupal/PHP Wordpress/PHP Django/Python Web design/graphics Apache administration in LAMP environment Please note that not every single skill is currently in demand; we are trying to get a feel for who knows what and hopefully anticipate future needs. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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voicebox email notifications on Debian with exim4
by Daniel-Constantin Mierla 12 Nov '12

12 Nov '12

Hello, I got into the crashing issue for freeswitch voicemail email notifications on Debian with exim4, reported at: * http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings My solution was to use msmtp alongside exim4 -- msmtp is used to relay the emails to the local exim4 instance. All seems to work ok, I wonder if anyone else is using same approach and got any issues or there are other solutions found meanwhile for debian+exim4 combination. I plan to put more details there, once the registration process for my account on the wiki is completed. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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Possible mod_smpp
by Michael Collins 08 Nov '12

08 Nov '12

Hello all, If you are interested in giving monetary support to have a professional software firm create mod_smpp then please contact me off list and I will give you more details. Thanks, Michael

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Attended transfer fails when in bypass or proxy mode
by Sergei 29 Oct '12

29 Oct '12

Good day all. I've recently installed freeswitch in our office and in general everybody is happy. The only disappointing thing - is attended transfer. I've tried to make it working on Cisco7960 and on Grandstream GXP2000 - but with no luck. Could please, anybody respond, who successfully makes attended transfer using bypass mode? In sip profile I use following options: <param name="media-option" value="resume-media-on-hold"/> <param name="media-option" value="bypass-media-after-att-xfer"/> Scenario is the simplest one: A calls B, B transfers to C (with attended transfer - like putting A on hold, making outcall to C, bridging two calls together). What I observe on Cisco phone - is following: 1. A calls B. B answers, puts A on hold and calls C. C answers and talks to B. Up to this time - everything is fine. 2. B presses 'transfer', in order to bridge two calls together. But, instead, C hangs up, A stays on hold and B looks like he just put A on hold. If B right now presses hold button - he will appear in dialog with A again. Did anybody observe such behavior? What could be the advice in such situation? Thank you in advance, Siarhei.

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Mod_limit questions
by ivdreg ivdreg 29 Oct '12

29 Oct '12

Hi All, I have some questions regarding mod_limit. 1. I'm using 16674 in production but tested and with 17001. I try mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using limit_hash never tries to write data to SQL. It works (counts sessions) on log screen but no data in MySQL/Sqlite. It seems that data is cached but cache never flushed to DB. Is anyone has this problem. My limits are: <action application="limit_hash" data="op_${sip_h_X-SRC-Cust} ${destination_number} 5 !INCOMING_CALL_BARRED" /> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust} ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" /> and they are applied on inbound call profile in same extension. 2. I need to implement rule maximum of 50 calls per day/24h from the same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in example "Limit calls to 5 calls every 10 minutes" is about concurrent call but not calls processed as count. Any idea ? Thanks for any help

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FS sends Hold signal to remote gateway
by Carlo Dimaggio 24 Oct '12

24 Oct '12

Hi all, I'm experiencing a strange behaviour when calls are on hold. Situation: Call 1. Phone A (internal profile) calls a number (PSTN endpoint) through a gateway on the external profile. If the phone sets on hold all is ok and the remote PSTN endpoint hears the FreeSWITCH music on hold. Call 2. PSTN endpoint calls through the same gateway the Phone A. If the Phone A sets on hold the call, FreeSWITCH sends an hold signal to the remote gateway, so there is no Music on hold. Is a normal behaviour? How I can have the same behaviour of call 1? Regards,

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mod_skypopen not generating inbound A-leg CDR
by DP . 22 Oct '12

22 Oct '12

Hi, Does mod_skypopen generate inbound A-leg CDRs? Simple scenario, inbound call from Skype to anywhere in Freeswitch (currently a Lua script, but also tried direct to extensions), does not generate any CDRs for that A-Leg. I tried transferring to another context, bridging, etc to no avail... only the b-leg, if present, generates a CDR. To eliminate my xml_cdr, I purposely set the post url to an invalid link, which throws a red crit error in console upon call hangup of regular mod_sofia calls. The mod_skypopen throws no error whatsoever, possibly indicating its not generating the CDR? Im running FreeSWITCH Version 1.0.head (git-5e4a514 2012-03-10 22-56-29 -0500). (Tried updating to the latest 1.2 branch last week but got random restarts and crashes of freeswitch, also kept getting "unclean" in the git version header after several make currents', had to quickly revert as this is a production box). Will clone the install again and try and test the latest git at a later date). Thanks, -Damian

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Freeswitch in server mode with OpenFire
by krunci 19 Oct '12

19 Oct '12

Hi, I'm new here on this forum, but I have some problem with configuring Freeswitch with OpenFire. First of all, did anybody have succes in configuring Freeswitch with OpenFire, and that it work properly?? Second, I have configure Freeswitch as a component, and add it in Openfire, but when I try to establish a call I have some problems: 1. When I call from SIP to XMPP (openfire) on Sip client have ringing all the time, but on XMPP client side I manage to answer the call, but on SIP side doesn't change anything. (ringing all the time) That means I coundn't establish any call. On Pidgin (XMPP side) is written "Call in progress". 2. In other way, when I try to call SIP client from XMPP client, on XMPP side is trying to call SIP side, but on SIP side doesn't change anything. Like nothing is happening on other side. So if someone who is expert in openfire and Freeswitch can help me with that problems? Thanks Regards, Krunci -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-in-server-mode-wit… Sent from the freeswitch-users mailing list archive at Nabble.com.

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xmpp
by Peter P GMX 19 Oct '12

19 Oct '12

I saw that xmpp is supported in Fresswitch. See wiki: http://wiki.freeswitch.org/wiki/Mod_xmpp_event Has anybody already set this up? I have found no mod_xmpp neither in my mod directory nor in the source? There was also a question: " Q: Is it possible to send commands to fs via xmpp? Answer: Yes. " Anybody knows what can be done here and how to do this? Best regards Peter

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sip profile - accept-blind-auth vs
by Daniel-Constantin Mierla 14 Oct '12

14 Oct '12

Hello, in the past I used to set: <param name="auth-calls" value="false"/> in the sip profile in order to skip user authentication for calls. Lately I started to play a bit with 1.2 stable branch and seems that setting auth-calls to false is no longer doing what I expected, calls being challenged for user authentication. Setting instead the accept-blind-auth to false got me what I wanted, like:  <param name="accept-blind-auth" value="true"/> But from the comment (checked the wiki as well, but has the same text) is a bit unclear what is the real purpose for it. Isn't auth-calls=false supposed to accept calls without user authentication anymore? For this particular case, I play some announcements, like 'user not available', and should work also for calls coming from outside. The access is restricted by IP address ACL, allowing SIP traffic only from my Kamailio instance. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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