- FreeSWITCH-dev - FreeSWITCH Mailing Lists

mod_conference tracking
by João Mesquita 28 Dec '10

28 Dec '10

Hello you all. I hope you are all having a good holiday. I have been looking for the best way of tracking conference usage for reporting purposes (basically answering the question: what happened and when on conference nº X?) and I thought of the following alternatives that I would like to discuss with you. 1. Using ESL and the CUSTOM events to keep track of what happened when (I guess that's the most widely used) 2. Creating a new entry on switch_caller_profile, that would be filled by the mod_conference module anytime something changes on the conference. Something similar to what the origination_caller_profile does but that would be manipulated by the mod_conference. This option could neatly integrate with the xml cdrs but would mess a little more on the core. Not sure it is desirable. 3. Make mod_conference spit it's own XML for each conference created so that we know what happens and can link offline to other cdr entities. Or, maybe I am completely dumb and trying to do something that it is not supposed to be done or forgetting some resource that is already available. João Mesquita

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Query related to enabling SRTP in FreeSWITCH-1.0.7
by Goutham BG 20 Dec '10

20 Dec '10

Hi, I have been trying to enable SRTP in FreeSWITCH-1.0.7 and have been facing some issues. I have the following entry in my dialplan XML file: *<extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="bridge" data="loopback/app=socket: 47.152.232.156:8084 async full"/> </condition </extension>* A SIP phone (Avaya 12XX) configured in "SRTP best effort" mode dials into this extension and is connected to the IVR. But the media is established in SRTP in one way and RTP in the other way. The phone offers the following SDP in the INVITE message: v=0 o=- 10170 10170 IN IP4 47.152.232.147 s=Sip Call c=IN IP4 47.152.232.147 t=0 0 m=audio 5016 RTP/AVP 0 8 18 101 102 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:102 X-nt-inforeq/8000 a=sendrecv m=audio 5016 RTP/SAVP 0 8 18 101 102 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:102 X-nt-inforeq/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjivoci1I/mVkt/Fq/ZsiY+ +ornJoXjZ5tSadho4 a=sendrecv As we can see, there are two "m=" lines in the SDP of the offer; one for RTP and another for SRTP. FreeSWITCH-1.0.7 answers the call by sending 200OK with the following SDP: v=0 o=FreeSWITCH 1291628984 1291628985 IN IP4 47.152.232.156 s=FreeSWITCH c=IN IP4 47.152.232.156 t=0 0 m=audio 11280 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=audio 0 RTP/SAVP 19 As you can see above, FreeSWITCH accepts the RTP stream and rejects the SRTP stream (by sending port as 0) in the SDP. The SIP phone sends the media in RTP(which is expected). But, FreeSWITCH sends the media in SRTP to the SIP phone. I believe this is a bug in FreeSWITCH as it is supposed to send the media in RTP since it accepted RTP in the answer (200OK). *Query: ======* In order to make FreeSWITCH select SRTP in the SDP of the answer(200OK), I made the following change(in *bold*) in FS dial plan: *<extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="set" data="sip_secure_media=true"/> <action application="bridge" data="loopback/app=socket: 47.152.232.156:8084 async full"/> </condition </extension>* In FreeSWITCH-1.0.6(before updating to 1.0.7), this worked and FS accepted the SRTP stream and rejected RTP in the answer(200 OK) as shown below: m=audio 0 RTP/AVP 19 m=audio 12084 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hgv7ClqDx1irTRrXq2NEm9Gbouw0969bBU3n+LcM But after updating the FreeSWITCH-1.0.6 to 1.0.7, the above mentioned dial plan change (i.e, setting sip_secure_media=true) is not working. It is still behaving in the same way as it did without the XML change. Can you please let me know if anything else needs to be added in dialplan XML file for enabling SRTP in this case in FreeSWITCH-1.0.7 or am I missing something here? I have referred the following FS wiki pages for making the SRTP changes: http://wiki.freeswitch.org/wiki/Secure_RTP http://wiki.freeswitch.org/wiki/SRTP Note: There is no issue when the SIP phone is configured in "SRTP only" mode where only SRTP stream is offered in the SDP of the INVITE. In this case, SIP phone and FreeSWITCH communicate properly using SRTP. This doesn't require setting "sip_secure_media=true" in the dialplan XML file. P.S: I am a newbie to FreeSWITCH. So, please forgive me if I am asking basic questions. Thanks Goutham B G

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Freeswitch & Solaris 10
by Marty Lee 20 Dec '10

20 Dec '10

Hi, just playing with building FreeSwitch on a Solaris 10 box and am hitting a link error when building the target '.libs/freeswitch' Undefined first referenced symbol in file make_mask32 freeswitch-switch.o ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch Looking at the source, make_mask32 seems to be defined in the spandsp/src/bit_operations.c file, which isn't involved in the linking at this point. Building with Sun Studio 12, just in case it's of any interest. I'll try and move the make_mask32 function in to the library and see if that cures the problem. Thought I'd ask here, as this kind of thing should appear on most platforms if it really is a problem - meaning it might be something at my end. m ----- Marty Lee e: marty(a)maui-systems.co.uk Technical Director v: +44 845 869 2661 Maui Systems Ltd f: +44 871 433 8922 Scotland, UK w: http://www.maui-systems.co.uk

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Jester pre-alpha released
by Chad Phillips -- Apartment Lines 19 Dec '10

19 Dec '10

Jester, the new Lua scripting toolkit for FreeSWITCH, is now officially released in the pre-alpha phase. Code is available in the freeswitch-contrib repository under hunmonk/jester. To learn more, you can start here: http://wiki.freeswitch.org/wiki/Jester

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overriding authentication scheme on mod_sofia
by Paulo Vicentini 18 Dec '10

18 Dec '10

Hi, I would like to override the scheme used for digest authorization Actually it is using the scheme coming from sip_www_authenticate_t in the sofia_reg_handle_sip_r_challenge function, ignoring scheme set in xml configuration I would like something like: if(gateway->register_scheme) scheme = gateway->register_scheme; before nua_authenticate(... I am patching sofia to accept HA1 Regards Paulo

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mod_python error
by peter_green lion 15 Dec '10

15 Dec '10

hi all, I see mod_python in wiki and i try to make some example as page: http://wiki.freeswitch.org/wiki/Examples_directory_py i have configure in dial plan as: <extension name="directory"> <condition field="destination_number" expression="^1234$"> <action application="python" data="directory"/> </condition> </extension> and in /usr/local/freeswitch/scripts/ add file directory.py as: import sys, time, sqlite3 from freeswitch import * digitpath = "/usr/local/freeswitch/sounds/en/us/callie/digits/8000/" custom_sounds_path = "/usr/local/freeswitch/sounds/custom/" def checkforgreeting(extension): conn = sqlite3.connect("/usr/local/freeswitch/db/voicemail_default.db") c = conn.cursor() c.execute("select name_path from voicemail_prefs where username=?", (extension,) ) # retrieve recorded_name path row=c.fetchone() console_log("alert", "row: %s\n" % (str(row))) c.close() if row[0]: return row else: return False def handler(uuid): alphabet = "abcdefghijklmnopqrstuvwxyz" numbers = "22233344455566677778889999" code_to_name = {} code = "" names = { "Allen, Larry": 1000, "Monroe, Beckey": 1001, } lnames = {} for name in names: lnames[name.lower()] = names[name] names = lnames def sayname(fullname): console_log("alert", "Now saying: " + fullname + "\n") # flip first and last... remove comma split_fullname = fullname.split(',') fname_lname = split_fullname[1].lstrip() + " " + split_fullname[0] # take the chars of a name and say each one session.execute("phrase", "spell," + fname_lname); # preprocessing before evaluating arg input # build codes from names dict for name in names: name3char = name[0:3] # empty out the code var code = "" for char in name3char: code = code + numbers[alphabet.index(char)] # code is the 3 digits code generated from the first 3 chars of the last name if not code in code_to_name: code_to_name[code] = [ name ] else: code_to_name[code].append(name) session = PySession(uuid) session.answer() session.execute( "sleep", "2000" ) digits_keyed = session.playAndGetDigits(3, 3, 10, 5000, "*#", custom_sounds_path + "dir-intro.wav", "", ""); # evaluate input # we want 3 digits console_log("alert", "digits_keyed: %s\n" % ( str(digits_keyed) )) if len(digits_keyed) == 3: # it must be in the code db generated from last names earlier if digits_keyed in code_to_name: console_log("alert", "Yes: %s\n" % (str(code_to_name[digits_keyed]))) if len(code_to_name[digits_keyed]) == 1: # only one extension matches console_log("alert", "Extension found: %s\n" % ( str(names[code_to_name[digits_keyed][0]]))) # transfer to the extension extension = str(names[code_to_name[digits_keyed][0]]) recorded_name = checkforgreeting(extension) if recorded_name: console_log("alert", "Saying recorded name\n") session.streamFile( str(recorded_name[0]) ) else: sayname(code_to_name[digits_keyed][0]) session.execute("phrase", "spell," + extension); session.execute( "sleep", "1000" ) # give option of if correct to press 1 otherwise * and start over digits_keyed = session.playAndGetDigits(1, 1, 3, 2000, "#", custom_sounds_path + "dir-instr.wav", "", "1|\*"); console_log("alert", "digits_keys: %s\n" % ( digits_keyed )) if digits_keyed == "1": session.transfer( extension, "XML", "default") else: if digits_keyed == "*": session.streamFile( custom_sounds_path + "dir-nomatch.wav" ) handler(uuid) # session.transfer( "777", "XML", "default") # session.hangup("1") else: # we matched more than one name for item in code_to_name[digits_keyed]: console_log("alert", "Found more than one extension: %s\n" % ( str(names[item]))) # say each one and give option of if not that one to continue console_log("alert", "item: %s\n" % (item)) extension = str(names[item]) recorded_name = checkforgreeting(extension) if recorded_name: session.streamFile( str(recorded_name[0]) ) else: sayname(item) session.execute("phrase", "spell," + extension); session.execute( "sleep", "1000" ) # give option of if correct to press 1 otherwise * and start over digits_keyed = session.playAndGetDigits(1, 1, 3, 2000, "#", custom_sounds_path + "dir-instr.wav", "", "1|\*"); console_log("alert", "digits_keys: %s\n" % ( digits_keyed )) if digits_keyed == "1": session.transfer( extension, "XML", "default") # session.hangup("1") session.streamFile( custom_sounds_path + "dir-nomore.wav" ) handler(uuid) # session.transfer("777", "XML", "default") # session.hangup("1") else: # no valid extension found so transfer back session.streamFile( custom_sounds_path + "dir-nomatch.wav" ) handler(uuid) # session.transfer("777", "XML", "default") # session.hangup("1") session.hangup("1") i log in soft phone and make call to :1234 ==========> call hangup and i see log in freeswitch -c: 2010-12-10 02:35:55.711484 [ERR] mod_python.c:200 Error calling python script TypeError: handler() takes exactly 1 argument (2 given) Traceback (most recent call last): File "<string>", line 1, in <module> NameError: name 'python_makes_sense' is not defined please suggest to fax it and make it work. another thing: if i have mysql database, how can i get value from mysql. example: i have table test in fusionpbx database. mysql> select * from test; +----+------+----------+-------------+ | id | user | password | ms | +----+------+----------+-------------+ | 1 | 123 | 12345678 | 87546345634 | +----+------+----------+-------------+ so: how can i get value password = 12345678 when i enter user =123 (in python code). thanks all so much

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FreeSWITCH Conference Call Today
by Michael Collins 15 Dec '10

15 Dec '10

Hello all, Here's today's agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_15 Chad Phillips is doing a follow up on Jester Mail and his Lua IVR toolkit. Talk to you all in an hour! -Michael

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ftmod_libpri - Channel Selection patch
by lakshmanan ganapathy 11 Dec '10

11 Dec '10

Hi all, When using ftmod_libpri, if a incoming call comes without "Channel Identification" ( in SETUP packet ), then the call didn't get proceeded. I've taken some code snippets from ftmod_isdn and I applied it in ftmod_libpri. I've tested it and in my setup it works fine. I've attached the patch here. Please verify it and found ok commit it. regards, Lakshmanan G.

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Scriptable test-suite
by Joegen E. Baclor 10 Dec '10

10 Dec '10

Hi, We are trying to work out a test suite for sipXecs media services and we thought that freeswitch itself is the best place to do it. I am initiallly thinking of controlling freeswitch using ESL inbound to simulate traversing of sipXecs media services ivr menus. However, although ESL has the most flexibility among the choices, I find the scripting modules really appealing specially because it exposes the ability to add new test scenarios. Am I correct that ESL inbound function(eg: placing independent outbound calls without having to hit a dialplan) is not currently doable via the mod_scripting_language plugins? Because of this, I am plannning to simply expose ESL inbound/outbound library via a google::v8 glue to make the test scenarios somehow scriptable. I am wondering if I am not crazy if i do so. I have came accross this thread old http://thread.gmane.org/gmane.comp.telephony.freeswitch.user/10926/focus=10… . I am wondeing if more work/thoughts came into this so i wont be reinventing the wheel. advice appreciated. Thanks! Joegen

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Asterisk migrations
by Michael Collins 10 Dec '10

10 Dec '10

Hey gang, I was just curious if anyone out there had created any tools to assist with migrating from Asterisk (or Trixbox, etc.) to FreeSWITCH. We're starting to see this scenario more and I would like to find out how our community members are handling it. Thanks, MC

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