April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to change Music-on-hold while digit_bind_action is used ?
by julien terrasson 24 Apr '13

24 Apr '13

Hi, First of all, i've been discovering freeswitch recently and i found that this is an awesome piece of software. I really enjoy discovering the possibilities of the software and creating scenarios. Thanks for all the people who made this possible and available. The reason i'm posting is that i need help me to guide me in troubleshooting.. This is the scenario i would like to acheive :* * *************************************************** * A & B are registered on FS * A call B. * When A press the star key => B is play a prompt that ask him if he would like to receive an email. => B can then choose to accept (by pressing "1") or reject (by pressing "2") Meanwhile, A is played a specific moh that tell him that he have to wait for B answer. This the way i tried to implement it : ****************************************** DIALPLAN : i'm using bind_digit_action with "*" to start a lua script on the *b_leg context*. => "bind_digit_action","my_context,*,exec:lua,send_email.lua,a,*peer*" EMAIL.LUA : *=> Change the moh on the A leg *api:executeString("uuid_setvar ".. a_leg_uuid .. " hold_music /usr/local/freeswitch/sounds/fr/ca/june/voxService/fr_wait_email.wav"); *=> Play and request digit (B leg)* digit = session:playAndGetDigits(1, 1, 1, 5000, '#', '/usr/local/freeswitch/sounds/fr/ca/june/voxService/fr_ask_email.wav', '', '\\d+'); 3/ Result : ************* When "*" is pressed for the 1st time, FS act as if moh wasn't change. It does only take into account the change the second time "*" is pressed. i assume thats moh change can only be taken into account when bind_digit_action is over. So i would like to know if there is any existing way to change this behaviour ? If not, what would be the best way of implementing this scenario ? Thanks for your answers. Julien

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Jira and the PasteBin... [Please READ ME]
by Ken Rice 24 Apr '13

24 Apr '13

Hey Guys, Please keep the following things in mind. Bugs should be reported via Jira. Yes its ok to ask on the mailing list to see if it you are just doing something wrong, but please open a bug on Jira if its a bug. A segfault is always a bug. (Even if you are doing it wrong, FreeSWITCH should not segfault) The Pastebin is a useful tool for sharing information between people, but please do not reference pastebin URLs in Jira Tickets. Pastebin entries are not guaranteed to survive for any period of time, and they can and will disappear at some point. This means of you open a ticket and you put a backtrace in the pastebin, if the pastebin is cleared so is your backtrace. (if you use the fscore_pb script to help collect the data, there is a download link right on the pastebin that will let you download a .txt file for that entry which you can then attach to the ticket ) Please Attach to jira ticket all backtraces, sip traces and logs as plain text files ( .txt files ) Also, please do not paste the results into the notes or the description of the issue. This just means we have to scroll down 97 screens of stuff to get the last notes on a ticket. These simple things will help out the developers and help make the bugs go away faster! Thanks! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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URGENT HELP! Customers cant make any calls -after MAKE CURRENT
by Sean Devoy 23 Apr '13

23 Apr '13

Hi All, I do truly hope someone responds before morning when my customers get back to work. I did a make current and it went swell! I rebooted, everything registered, looked great - BUT . Everything everyone dials gets codec mismatches and then [INCOMPATIBLE_DESTINATION] The codecs look like this: 2013-04-22 23:42:02.138488 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] But I don't know how to set that parameter in in FS or on the devices! Any help greatly appreciated. Sean

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FreeSWITCH Weekly News and Notes
by Michael Collins 23 Apr '13

23 Apr '13

A belated welcome to the week of April 22! News and notes is a day late because I spent all day yesterday working on the new FreeSWITCH book <http://www.packtpub.com/freeswitch-1-2/book>. Good news - we have submitted all the content! It's now just a matter of getting all the edits and reviews done. We're a little more than halfway through that process. We'll update you as more news becomes available. Last week we had a conference call that<http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_17>lasted quite a while. Dave Kompel showed us some tricks with using MS Visual Studio 2012 to build FreeSWITCH applications. That discussion dovetailed into a more general discussion about some of the other things Dave has been doing with FreeSWITCH under Windows. In the weeks and months ahead, Dave will be doing some more Windows-only discussions outside of the main weekly conference call. Keep an eye on the mailing list<http://lists.freeswitch.org>for more announcements. Tomorrow we have scheduled an engineer from JeraSoft<http://www.jerasoft.net/products/vcs/index.html>who will be talking about their routing and billing solution that has been tightly integrated with FreeSWITCH 1.2. If you have FreeSWITCH in a production environment where billing and routing are core elements then this discussion <http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_24> will definitely be of interest. Have a great rest of the week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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http_cache with mod_httapi
by Srigo 23 Apr '13

23 Apr '13

Hi, I want to use http_cache to play a file in httapi dialplan but its not working. Here is what im getting. Here is my httapi xml: <document xmlns:xs="http://www.w3.org/2001/XMLSchema" type="xml/freeswitch-httapi"> <params/> <variables> <dtmf>null</dtmf> </variables> <work> <execute application="export" data="app_id=${app_id}"/> <execute application="log" data="DEBUG: Matching calendar: opening_hours, status:open"/> <execute application="flush_dtmf"/> <execute application="log" data="INFO: application=ivr;value=http://xxxxxxx.com:8080/tel/media_srv/ib_fr_wel.wav;user_entry=null;max_failure=0;loop=0;app_id=593924623;story_id=1;status=start"/> <execute application="set" data="${http_prefetch(http://xxxxxxx.com:8080/tel/media_srv/data/iBrowse/ib_fr_wel.wav)}"/> <execute application="set" data="file_cached=${http_get(http://xxxxxxx.com:8080/tel/media_srv/data/iBrowse/ib_fr_wel.wav)}"/> <execute application="set" data="app_id=593924623"/> <execute application="set" data="story_id=1"/> <playback name="user_entry" file="${http_get(http://xxxxxxx.com:8080/tel/media_srv/data/iBrowse/ib_fr_wel.wav)}" error_file="" digit-timeout="1500" input-timeout="2000"> <bind strip="#">~\d+</bind> <bind strip="#">~\d+</bind> <bind strip="#">~\d+</bind> </playback> </work> </document> Here is what im getting in logs: 2013-04-23 14:40:59.234148 [INFO] switch_ivr_async.c:194 Digit parser HTTAPI: Setting realm to 'default' 2013-04-23 14:40:59.234148 [DEBUG] switch_ivr_async.c:285 Digit parser HTTAPI: binding \d+/default/0 callback: 0x7fd528bf0840 data: 0x7fd524046640 2013-04-23 14:40:59.234148 [DEBUG] switch_ivr_async.c:285 Digit parser HTTAPI: binding \d+/default/0 callback: 0x7fd528bf0840 data: 0x7fd524046680 2013-04-23 14:40:59.234148 [DEBUG] switch_ivr_async.c:285 Digit parser HTTAPI: binding \d+/default/0 callback: 0x7fd528bf0840 data: 0x7fd5240466c0 2013-04-23 14:40:59.234148 [INFO] switch_ivr_async.c:194 Digit parser HTTAPI: Setting realm to 'default' 2013-04-23 14:40:59.234148 [ERR] switch_core_file.c:112 Unknown file Format [$] Thank you -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/http-cache-with-mod-httapi-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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PTIME 0.020 vs 0.030 and call quality
by Sean Devoy 23 Apr '13

23 Apr '13

Hi all, Does call quality improve when ptime is shortened? I have seen posts saying 40ms resulted in choppy audio. Clearly FS likes 0.020. I am trying to decide if I should go through and update my Cisco and Linksys devices to use 0.020. Thanks, Sean

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a way to disable hangup_after_bridge when an attended transfer happens?
by Michel Brabants 23 Apr '13

23 Apr '13

Hello, is there an elegant way to disable hangup_after_bridge when an attended transfer happens? Thanks, Michel

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Compilation Error In Freeswitch
by Arpit Gadle 23 Apr '13

23 Apr '13

Dear All, I am getting this compilation error on one of my box oma_ss7_la-ftmod_sangoma_ss7_cfg.o src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c: In function 'ftmod_ss7_mtp2_dlsap_config': src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:878: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:879: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:880: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:881: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:882: error: expected expression before ';' token src/ftmod/ftmod_sangoma_ss7/ftmod_sangoma_ss7_cfg.c:883: error: expected expression before ';' token Thanks & Regards, Arpit Gadle Mobile : +91-9303964208

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calls dropped after refer with replaces
by Michel Brabants 23 Apr '13

23 Apr '13

Hello, when I send a refer with replaces from my polycom, the calls are dropped with which I'm trying to do an attended transfer. When leg arrives in CS_PARK and the other in CS_HIBERNATE. An unbridge happens afterwards on the leg that sent the refer-replaces, but in the end all legs are dropped. Any idea why this could be happening? I tried to look at the sourcode, but I couldn't find my problem yet. Thanks, Michel

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freeswitch not replying on port according to srv-record
by Michel Brabants 23 Apr '13

23 Apr '13

Hello, one more question. While performing tests, I noticed that freeswitch didn't respond to the port that is specified in the srv-record, which is in the contact-address. This is when it receives an incoming invite in one of the defined gateways in freeswitch. The srv-record has 5062 in it, but freeswitch doesn't seem to try to resolve the srv-record. Kind regards, Michel

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