November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Explanation of rtp_audio_in / out fields in XML CDR's?
by Mick Stevens 05 Dec '12

05 Dec '12

Hi Folks, I'm trying to use FS XML CDR's to diagnose historic audio problems. I think I can work out some of the rtp_audio_in / out fields (raw bytes & media bytes being nearly equal looks like a good sign) but am wondering about the skip, cng & flush fields for example? I have tried Googling this & can find evidence of other people having asked this question but not of the answer. I have also checked my FS 106 & Cookbook book's without success. The wiki appears to be down at the moment so my apologies if the answer lies there. I know how to do this in real time using wireshark etc but am interested in being able to do some analysis on historic problems reported by customers that aren't willing/able to replicate the problem in order for a protocol trace to be captured. Any help much appreciated! Rgds, Mick Tel/SMS. +44(0)7967 594432 Fax. +44(0)7053 452429 Email/IM. mickstevens(a)yahoo.com Skype: mick_stevens www.facebook.com/mickstevens www.twitter.com/mickstevens

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Can not use Python ESL with Django and apache web server
by chaiyawut.so 05 Dec '12

05 Dec '12

I tried freeswitch python ESL library with Django and apache mod_wsgi today. I already compiled and installed pymod in Freeswitch source folder. My freeswitch server runs as root and my DJnago web server runs as www-data and mod_wsgi. When used command "ESLconnection("localhost", "8021", "ClueCon")" in my views.py to connect to freeswitch server, django froze permanently and it said waiting for the request... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-not-use-Python-ESL-with-D… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch is crashing when httapi playback application is executed
by Srigo 04 Dec '12

04 Dec '12

Hello, I'm using Httapi for dynamic application service. The problem, I'm getting is when im using httapi with the application "playback", freeswitch is crashing. Seems the crashed is caused by the parameter of "file" in playback. Thats mean when i'm getting a file from a remote server (http://xxxxxx.com/test.wav) its crashing but when using a local wav file everthing is fine. here is the httapi xml: <document type="xml/freeswitch-httapi"> <params/> <variables> <sou>someval</sou> </variables> <work> <execute application="flush_dtmf"/> <execute application="set" data="${http_prefetch(http://xxxxxxx.com/test.wav)}"></execute> <execute application="info"/> <playback name="user_entry" file="http://xxxxxxx.com/test.wav" loops="3" digit-timeout="1000" input-timeout="1000"> <bind strip="#">~\d+</bind> </playback> </work> </document> Here the log before the freeswitch crash: freeswitch: src/switch_event.c:710: switch_event_get_header_ptr: Assertion `event' failed. I try to use http_cache to try to resolve the prob: <playback name="user_entry" file="${http_get(http://xxxxxxx.com/test.wav)}" loops="3" digit-timeout="1000" input-timeout="1000"> <bind strip="#">~\d+</bind> </playback> but i'm getting the following error in the log: 2012-11-23 19:06:10.015936 [ERR] switch_core_file.c:112 Unknown file Format [$] Finally i try the following idea: - set a variable (<execute application="set" data="play=${http_get(http://xxxxxxx.com/test.wav)}"></execute> then use in playback parameter the variable ${play} but the variable is not parsed! 2012-11-23 17:30:09.855933 [ERR] mod_native_file.c:74 Error opening /usr/share/freeswitch/sounds/en/us/callie/${play}.PCMA Could anyone please tell me where i did the mistake ? thx a lot -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-is-crashing-when-h… Sent from the freeswitch-users mailing list archive at Nabble.com.

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WebRTC Call
by Ros P 04 Dec '12

04 Dec '12

Hello, Has anyone had any luck in implementing a webrtc call? I have been able to register using the webrtc2sip gw and sipml5 client but no audio goes through. As this seems to be a client problem (issue 39<http://code.google.com/p/sipml5/issues/detail?id=39>), does anyone have any other suggestions? Anything successfully tested? Kind Regards, ros

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Questions about RTMP endpoint
by Germán Ruiz 04 Dec '12

04 Dec '12

Hi, I'm adding voice calls to a web site. I'm using RTMP endpoint and Flex clientprovided. I have the following questions: - Can flex clients (using RTMP protocol) register their presence and are visible by others connected using SIP? - Is it possible to send and receive text messages with RTMP protocol? Thanks Germán

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mod_cdr_mongodb follows json rfc4627?
by Jose Fco. Irles Durá 03 Dec '12

03 Dec '12

Hello everybody, first of all, sorry for my english. I have a test server with the mod_cdr_mongodb backend configured. I'm trying to parse the cdrs with the mongodb driver for java, but i have a trouble with a part of the json that freeswitch sends (mongo saves the json document without errors). I haven't an API function to extract some data. Firstly I thought that the problem was in the java driver but I'm not sure. The problem is in the "callflow" part, with the json standard[1], a json document can't have two keys with the same name, but mod_cdr_mongodb builds the json with a 'n' callflow objects. Also it happends inside callflow object, with origination, originator and originatee objects. mod_cdr_mongodb saves the json with this format: http://pastebin.com/VRz6s0eb I modified the source of the backend adding json arrays around this objects and now I can parse without problems. The output with the modifications: http://pastebin.com/v05FN9Ta I'm in the correct way or I'm missing something? [1] http://www.ietf.org/rfc/rfc4627.txt Section 2.2 Regards -- Jose Fco. Irles Durá

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XMPP Presence?
by Gavin Henry 03 Dec '12

03 Dec '12

Hi all, We're running our own XMPP chat server and would like to tie this in to FreeSWITCH. Has anyone tested presence via XMPP? The wiki is very light on this and would be happy to update. Thanks.

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mod_db help needed
by Nitin Tomer 02 Dec '12

02 Dec '12

Hi, I am trying to enter some data (entered by caller, received by Play and Get Digits). But I am not able to get much information about it from mod_db wiki. I want to enter data in an Oracle database. Please tell me what all I need to do, in order to do this. Regards Nitin Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL) accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.

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Caller Name with FreeTDM libpri
by Spencer Thomason 30 Nov '12

30 Nov '12

Hello, I have an ISDN PRI where the caller name is sent in a FACILITY message after the initial SETUP message. Does anyone know how I can access this? I'm using FreeTDM in DAHDI mode with libpri. Thanks, Spencer

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moh
by Jason Holden 30 Nov '12

30 Nov '12

Is there a way that moh is always playing in the background so when I place a user on hold they are not always starting at the beginning of the hold file?

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