July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Compile and install mod_cdr
by Rodrigo P. Telles 11 Jul '08

11 Jul '08

Hi Folks, I'm trying to compile and install mod_cdr following the wiki page http://wiki.freeswitch.org/wiki/Mod_cdr but it seems obsolete and broken. BTW I try do add "event_handlers/mod_cdr" on modules.conf and it builds fine. But when I try to run freeswitch it complains with: 2008-05-09 11:35:03 [CRIT] switch_loadable_module.c:727 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_cdr.so **/usr/local/freeswitch/mod/mod_cdr.so: undefined symbol: _ZN12CDRContainer7add_cdrEP19switch_core_session** Does some one here build mod_cdr sucessfully? If so, please share the steps with us. Thanks in advance, Rodrigo Telles

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Directed call pickup
by whudson05 11 Jul '08

11 Jul '08

Hi, I've been looking at using FreeSwitch for the last few days as a PBX. One of the features I'm quite keen on having is directed call pickup. In other words answering an extension that is currently ringing by dialing a code and then the extension number. So far I don't seem to have come across anything about this. Is it possible currently possible or would it require writing a new module? Thanks -- View this message in context: http://www.nabble.com/Directed-call-pickup-tp18385751p18385751.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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bridge and playback
by Mihail Krivushin 11 Jul '08

11 Jul '08

I want to realise next scheme: 1. try call (bridge) to local extensions 2. if no one answer in 20s playback wav with attention to wait 3. bridge to mobile phone I dont understand - I must set <action application="set" data="hangup_after_bridge=false"/>? If I set this and leg b hangup (dont answer or normal clearing) playback will execute? How I can do 2&3 only on NO_ANSWER cause? -- С уважением, Кривушин Михаил Евгеньевич Системный администратор ЗАО "СОЮЗ Т" тел. +7 913 865 7 866 jid: mkrivushin(a)souzt.com icq: 218 744 127

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silence, non-silence or tone detection
by Julien Chavanton 10 Jul '08

10 Jul '08

Hi, According to what I have found in the wiki, these features are still not available in Freeswitch ?

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IVR: dynamic call transfer/bridge
by Birgit Arkesteijn 10 Jul '08

10 Jul '08

Hi, I'm looking at the various ways to implement IVR in FreeSWITCH. My dialplan is generated dynamically, using xml_curl. What I'd like to achieve is: - user rings a number - FreeSWITCH detects the outing number and asks "You've reached XXXX in YYYY. If you want to continue press 1. Alternatively this call will terminate." When the user presses '1', the call has to be transferred to a dynamic number. Can I (and if so, how do I) pass the (dynamic) extension (actually bridge to gateway) from my dialplan to the IVR? ivr.conf.xml seems easy to use. Can I use dialplan variables in ivr.conf.xml, or do I need to use lua or js? I know I can use <action application="set" data="to_be_transferred=XXX"/> Would I be able to use this new variable in ivr.conf.xml or any of the other IVR tools? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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freeswitch: su_alloc.c:800 su_free: `Assertion allocation' failed.
by Faraz R. Khan 10 Jul '08

10 Jul '08

The above error has happened twice in the last three days (since I have playing with XML CDRs). Causes freeswitch to crash. It generated a core dump the first time but not the second time (The entire xen Dom0 crashed with it!). I will report back with a core dump the next time it happens. My system configuration is: Xen DomU- Ubuntu Hardy 8.04 with 256 Mb Ram X86_64 Xen Dom0- Ubuntu Hardy 8.04 with Xen 3.2 kernel on both is 2.6.24 Freeswitch compiled with --mod-python and -enable-64 I understand the above is incomplete info but I was just wondering if its a known issue or not and if somebody can simply tell me what can cause this problem. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz

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More Sample TTS Applications
by Tim Hunt 10 Jul '08

10 Jul '08

I noticed that mod_flite has just been added to the development trunk. As it turns out, I had been experimenting using the external flite executable, since every switch needs to be able to say the time and the weather forecast .... Anyway, it was a trivial change to start using mod_flite. For anybody that's interested, the resulting code can be seen at:- http://www.timhunt.net/wiki/FreeSwitch:Main Enjoy Tim Hunt tim(a)timhunt.net

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Looking for paid help to set up Freeswitch as an SBC
by Matt Darnell 10 Jul '08

10 Jul '08

This is what I am trying to accomplish: SIP -> public internet -> freeswitch -> private network -> exchange2007 Help with this would be compensated vie rentacoder, paypal, etc. All the software is setup, just need to do the configs. -Matt

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mod_conference "auto-record" enhancement
by Chris Danielson 09 Jul '08

09 Jul '08

FreeSWITCH may or may not need this but I added an auto-record feature into the mod_conference module. Essentially, one can specify within the "conf/autoload_configs/conference.conf.xml" the following attribute: <param name="auto-record" value="/var/myNFSshare/audio/" /> When this is specified the conference will automatically record the conference->name with an appended ".wav" into that folder. I built this out of necessity for my current implementation. I thought since it is able to be toggled off and on based on specifying the "auto-record" parameter that it might be worthy of inclusion. Attached is my svn diff. Kind Regards, Chris -- *Chris Danielson* Software Consultant and Co-Founder Web: MaxPowerSoft, LLC <http://www.maxpowersoft.com/> Email: chris(a)maxpowersoft.com <mailto:chris@maxpowersoft.com> Index: mod_conference.c =================================================================== --- mod_conference.c (revision 8910) +++ mod_conference.c (working copy) @@ -27,8 +27,8 @@ * Neal Horman <neal at wanlink dot com> * Bret McDanel <trixter at 0xdecafbad dot com> * Dale Thatcher <freeswitch at dalethatcher dot com> + * Chris Danielson <chris at maxpowersoft dot com> * - * * mod_conference.c -- Software Conference Bridge * */ @@ -230,6 +230,7 @@ char *caller_id_number; char *sound_prefix; char *special_announce; + char *auto_record; uint32_t max_members; char *maxmember_sound; uint32_t anounce_count; @@ -880,6 +881,14 @@ switch_mutex_lock(globals.hash_mutex); globals.threads++; switch_mutex_unlock(globals.hash_mutex); + + if (conference->auto_record) { + char *rfile = switch_mprintf("%s%s%s", conference->auto_record, conference->name, ".wav"); + switch_assert(rfile); + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Auto recording file: %s\n", rfile); + launch_conference_record_thread(conference, rfile); + switch_safe_free(rfile); + } while (globals.running && !switch_test_flag(conference, CFLAG_DESTRUCT)) { switch_size_t file_sample_len = samples; @@ -4806,6 +4815,7 @@ switch_status_t status; int comfort_noise_level = 0; char *suppress_events = NULL; + char *auto_record = NULL; /* Validate the conference name */ if (switch_strlen_zero(name)) { @@ -4923,6 +4933,8 @@ } } else if (!strcasecmp(var, "suppress-events") && !switch_strlen_zero(val)) { suppress_events = val; + } else if (!strcasecmp(var, "auto-record") && !switch_strlen_zero(val)) { + auto_record = val; } } @@ -5078,6 +5090,10 @@ clear_eflags(suppress_events, &conference->eflags); } + if (!switch_strlen_zero(auto_record)) { + conference->auto_record = switch_core_strdup(conference->pool, auto_record); + } + /* caller control configuration chores */ if (switch_ivr_digit_stream_parser_new(conference->pool, &conference->dtmf_parser) == SWITCH_STATUS_SUCCESS) {

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XMLCDR Question: <callflow profile_index="x">
by Michael Collins 09 Jul '08

09 Jul '08

I am experimenting with XML CDRs and I was curious about this part of the record. It seems that with each transfer to a different extension I'm getting a new callflow profile. That's good - I just want to make sure that my understanding of this is correct. Will a new callflow profile get generated for each successive transfer to a different extension? Also, are there other scenarios that generate a new callflow profile? Lastly, is there a way to force a new callflow profile to occur from within the dialplan? (That last one is just a curiosity, although it would help me handle some quirky CDR needs with the IVR that I'm working on.) Thanks! -MC P.S. - as per usual, any info provided here will go straight to the wiki!

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