August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Per-channel fsctl loglevel?
by Avi Marcus 11 Oct '12

11 Oct '12

Hi. I've been developing a script for someone to handle credit card transactions via IVR. Is there a way to just set the particular channel/call to fsctl loglevel 6, rather than the entire switch..? (dtmf is shown in /log 7) And.. similarly is there a way to blank out the var digits_dialed in the xml_cdr, from within FS, before the end of the call? Thanks! -Avi

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Pause in bridge
by Systeembeheer 01 Oct '12

01 Oct '12

Hi, I'm trying to configure the following: When a call comes in, it is first bridged to one extension and after 10 seconds to a second extension. It is possible to issue the bridge command with a timeout and continue settings to one extension and a second bridge command to both extensions. There's some problems with that: - The first extension can answer and 'just miss' the first bridge attempt. - If the first call is cancelled by the user, it rings again 10 sec. later. Is there a way to bridge to an extension and bridge to a second extension 10 sec. later without breaking the first bridge? Or bridge to extensions with a pause of 10 sec. for one of them? Thanks! steveN

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Saving phone information from a Polycom IP501
by Todd Bailey 27 Sep '12

27 Sep '12

Just wondering if it's possible to download data from a phone and save it into separate files or into a database? I just save a number into the phone and it occurred to me the minute I have to reboot or power cycle the phone, all contact info will be lost. If possible, I'd like to store the contact information and call history into a database for future reference, and perhaps being able to store that information into my email and contact management application (currently Evolution)

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FreeSWITCH & Heartbeat / Pacemaker - Please Help setting up
by Avi Marcus 26 Sep '12

26 Sep '12

Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far, no need to load balance.) I already have heartbeat able to grab the public IP if the first box goes down, but since I'm basically completely unfamiar with pacemaker, I'd like some help. I'd imagine I'm not the only one doing this, there should be no reason for me to recreate the wheel. Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA Specifically: 1) It seems freeswitch takes 8 seconds to restart. So if it crashes (unlikely) calls should immediately get shunted over to the other box. How do I set that up? 2) How do I automatically trigger a sofia recover? I know via commanline it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that? And I suppose a reload_xml before (maybe after?) is a good idea, too. Also, can I put <param name="track-calls" value="true"/> in a global file, or does it need to be in the actual profile files? And anything else I may be overlooking. Thanks guys! -Avi Marcus

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