February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Auto-nat affects all profiles?
by Dmitry Bely 23 Feb '11

23 Feb '11

I have enabled auto-nat only for the external profile (port 5080), but FreeSWITCH also maps SIP port for the internal one: freeswitch@internal> show nat_map port,proto,proto_num,sticky 5060,udp,0,0 5060,tcp,1,0 5080,udp,0,0 5080,tcp,1,0 4 total. Is this intended or just a bug? - Dmitry Bely

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Problems getting asterisk registered with FS sbc
by Johannes Jakob 23 Feb '11

23 Feb '11

Fellow FreeSWITCH Admins, I'm having a hard time, getting a Trixbox 2.8 box to register with our FreeSWITCH SBCs. The FreeSWITCHes are running FreeSWITCH-mod_sofia/1.0.head-git-7847289, the asterisk on the trixbox is Asterisk 1.6.0.22-samy-r60. The user's directory entry looks like this: <include> <user id="748732" cidr="10.16.153.163/32"> <params>  <param name="t38-passthru" value="true"/> </params> <variables> <variable name="client_asserted_identity" value="03023671836"/> <variable name="accountcode" value="748732"/> <variable name="contract" value="2253"/> <variable name="nibble_rate" value="1"/> <variable name="nibble_account" value="2253"/> <variable name="proxy_media" value="true"/> <variable name="bypass_media" value="false"/> <variable name="user_context" value="mysip.net"/> <variable name="default_language" value="de"/> <variable name="language" value="de"/> </variables> </user> </include> Asterisk's register string: 748732@mysip.net@sbc1.mysip.net/748732 I'm getting the "normal" FS errors for wrong credentials: 2011-02-22 18:03:57.484939 [WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [748732(a)mysip.net] from ip 10.16.153.163 2011-02-22 18:03:57.491471 [WARNING] sofia_reg.c:1204 SIP auth failure (REGISTER) on sofia profile 'internal' for [748732(a)mysip.net] from ip 10.16.153.163 but why am I getting these? I specified the right address in the cidr statement! Why is it even bothering with anything else but the right user@domain and IP-address? There are some other asterisk boxes (> 1.8.2) registering to this SBC with equal settings just fine, what's wrong with this little trixbox system? ;) Of course I did get you some SIP traces as well: 18:00:37.063410 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 419 E`..f...>.7.^...^..B.......-REGISTER sip:mysip.net SIP/2.0 Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK3e70680b;rport Max-Forwards: 70 From: <sip:748732@mysip.net>;tag=as77c8852d To: <sip:748732@mysip.net> Call-ID: 53d04cc277cfe60301bddb6d79033420(a)10.16.153.163 CSeq: 102 REGISTER User-Agent: Asterisk PBX 1.6.0.22-samy-r60 Expires: 1800 Contact: <sip:748732@10.16.153.163> Event: registration Content-Length: 0 18:00:37.074085 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 657 E...F...?.Vc^..B^...........SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK3e70680b;rport=5060 From: <sip:748732@mysip.net>;tag=as77c8852d To: <sip:748732@mysip.net>;tag=5jD9Qcg3N9S6p Call-ID: 53d04cc277cfe60301bddb6d79033420(a)10.16.153.163 CSeq: 102 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="mysip.net", nonce="ce2bccbf-a27b-43c8-b7b0-a89ab429d8a7", algorithm=MD5, qop="auth" Content-Length: 0 18:00:37.074969 IP 10.16.153.163.5060 > 10.16.133.66.5060: UDP, length: 672 E`..f...>.6.^...^..B........REGISTER sip:mysip.net SIP/2.0 Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK30df5010;rport Max-Forwards: 70 From: <sip:748732@mysip.net>;tag=as03431ba4 To: <sip:748732@mysip.net> Call-ID: 53d04cc277cfe60301bddb6d79033420(a)10.16.153.163 CSeq: 103 REGISTER User-Agent: Asterisk PBX 1.6.0.22-samy-r60 Authorization: Digest username="748732(a)mysip.net", realm="mysip.net", algorithm=MD5, uri="sip:mysip.net", nonce="ce2bccbf-a27b-43c8-b7b0-a89ab429d8a7", response="133a0ba843fe9f5afba67d1377fa8c11", qop=auth, cnonce="119cf18c", nc=00000001 Expires: 1800 Contact: <sip:748732@10.16.153.163> Event: registration Content-Length: 0 18:00:37.081517 IP 10.16.133.66.5060 > 10.16.153.163.5060: UDP, length: 532 E..0F...?.V.^..B^.........1.SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.16.153.163:5060;branch=z9hG4bK30df5010;rport=5060 From: <sip:748732@mysip.net>;tag=as03431ba4 To: <sip:748732@mysip.net>;tag=6U61S706jjgSj Call-ID: 53d04cc277cfe60301bddb6d79033420(a)10.16.153.163 CSeq: 103 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 Can somebody point me in the right direction? Thanks and best regards, John

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RTP Proxy config and DTMF conversion problem
by Mac 23 Feb '11

23 Feb '11

Dear ALL, Thats my first post here. I am struggling for some time with DTMF issue. Let me introduce you my configuration. The main task is to configure RTP Proxy with full topology hiding - OPERATOR_1 ------- Freeswitch ------- OPERATOR_2 Here is a prt of my config: - sip profile. I have one new sip profile with two gateways (OPERATOR_1 and OPERATOR_2) <include> <gateway name="OP1"> <param name="username" value="X1.Y1.V1.Z1"/> <param name="password" value="09321234"/> <param name="proxy" value="X1.Y1.V1.Z1"/> <param name="from-domain" value="test.com"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="extension" value="test"/> <param name="extension-in-contact" value="true"/> <param name="outbound-codec-prefs" value="$${outbound_codec_ prefs}"/> <param name="disable-transcoding" value="true"/> <param name="inbound-late-negotiation" value="true"/> </gateway> </include> - dialplan <include> <context name="public"> <extension name="term"> <condition field="network_addr" expression="^(X1\.Y1\.V1\.Z1|X2\.Y2\.V1\.Z1)$" break="on-false"/> <condition field="destination_number" expression="^46(\d{9})$"> <action application="set" data="proxy_media=true"/> <action application="set" data="bypass_media=false"/> <action application="start_dtmf"/> <action application="bridge" data="sofia/gateway/OP1/$0"/> </condition> </extension> </context> </include> Everything is fine, but i have problem with DTMF conversion from RFC2833 to inband. Refering to http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmfand http://wiki.freeswitch.org/wiki/Proxy_Media i have done all the necessary things. vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,G.729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,G.729,PCMU,GSM"/> The after-effect is following output: [ERR] switch_core_io.c:1055 Codec PROXY PASS-THROUGH encoder error! Placing <param name="dtmf-type" value="none"/> does not help anyway. Could sb point me where the problem is located? Thanks in advance, Mac

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Re: [Freeswitch-users] mod_odbc_query
by Peter Olsson 23 Feb '11

23 Feb '11

Rebuild FS with ODBC support, you were probably missing some ODBC devel files when configuring and building FS. /Peter ----- Reply message ----- Från: "Sam" <u2nsam(a)gmail.com> Datum: ons, feb 23, 2011 07:48 Rubrik: [Freeswitch-users] mod_odbc_query Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> I am getting this error while loading mod_odbc_query 2011-02-23 12:01:58.487072 [ERR] mod_odbc_query.c:489 No core ODBC available! 2011-02-23 12:01:58.487072 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/fs/novanet/mod//mod_odbc_query.so **Module load routine returned an error** Regds Sam On Mon, Feb 21, 2011 at 6:36 PM, Avi Marcus <avi(a)avimarcus.net<mailto:avi@avimarcus.net>> wrote: Like the wiki says, you need to pull the git CONTRIB. There's a link in that wiki page.. -Avi On Mon, Feb 21, 2011 at 2:52 PM, Sam <u2nsam(a)gmail.com<mailto:u2nsam@gmail.com>> wrote: Do i need to pull down git for it ? regds Sam On Mon, Feb 21, 2011 at 3:47 PM, Avi Marcus <avi(a)avimarcus.net<mailto:avi@avimarcus.net>> wrote: You can find the download / compile instructions on the wiki: http://wiki.freeswitch.org/wiki/Mod_odbc_query <http://wiki.freeswitch.org/wiki/Mod_odbc_query>-Avi On Mon, Feb 21, 2011 at 6:30 AM, Sam <u2nsam(a)gmail.com<mailto:u2nsam@gmail.com>> wrote: Hello, How to download mod_odbc_query source, there is not such source in 1.0.6 or 1.0.7 . Regds Sam _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4d64ad4632761097013501!

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mod_odbc_query
by Sam 23 Feb '11

23 Feb '11

Hello, How to download mod_odbc_query source, there is not such source in 1.0.6 or 1.0.7 . Regds Sam

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Message Playback Stops
by Michael De Lorenzo 23 Feb '11

23 Feb '11

Hello, I have a Freeswitch installation that is intended to make many calls (thousands) and playback a single wav file. The calls are successfully processed (the recipient's phone rings), but the call almost immediately disconnects, after about 1s. Everything seems to work fine if I'm only pushing one or two calls through the Freeswitch instance, but as soon as I turn up the call rate (I'm still only doing about 50 concurrent sessions) the playback begins to fail. I've watched the calls go out from the console and nothing looks out of the ordinary, except that the calls are disconnected with NORMAL CLEARING prior to completion. Here's the Lua script I'm using... profile_id = argv[1]; account_code = argv[2]; client_id = argv[3]; caller_id_name = argv[4]; caller_id = argv[5]; dial_id = argv[6]; number_to_call = argv[7]; message_to_play = argv[8]; max_retries = argv[9]; retry_interval = argv[10]; local human_detected = false local voicemail_detected = false; local message_played = false; recordings_directory = "/usr/local/freeswitch/recordings/messages/"; function setDialVariables(set_as_session_variables) local s = "profile_id=" .. profile_id; s = s .. ",account_code=" .. account_code; s = s .. ",client_id=" .. client_id; s = s .. ",caller_id_name=" .. caller_id_name; s = s .. ",caller_id=" .. caller_id; s = s .. ",dial_id=" .. dial_id; s = s .. ",number_to_call=" .. number_to_call; s = s .. ",message_to_play=" .. message_to_play; freeswitch.consoleLog("notice", s .. "\n"); return s end function printSessionVariables() freeswitch.consoleLog("notice", "******* PRINTING SESSION VARIABLES **********\n"); -- ommitted freeswitch.consoleLog("notice", "**********************************************\n"); end function onInput(s, type, obj, arg) if(type == "event" and voicemail_detected == false) then freeswitch.consoleLog("debug","************ VOICE MAIL/ANSWERING MACHINE DETECTED *************\n"); voicemail_detected = true; return "break"; end return true; end function playbackMessage(sleepTime) message_played = false; session:sleep(sleepTime); -- play a file message_file = recordings_directory .. message_to_play; freeswitch.consoleLog("notice", "Playing file: " .. message_file .. "\n"); session:streamFile(message_file); freeswitch.consoleLog("notice", "!!!!! Finished playing the file !!!!!\n"); message_played = true; end session = freeswitch.Session("{" .. setDialVariables(false) .. ",ignore_early_media=true,origination_caller_id_name=" .. caller_id_name .. ",origination_caller_id_number=+1" .. caller_id .. "}sofia/gateway/gateway_" .. profile_id .. "/" .. number_to_call); while(session:ready()) do setDialVariables(true) session:answer(); -- session:execute("continue_on_fail","true"); session:setInputCallback("onInput","true"); session:execute("avmd","start"); playbackMessage(200); vm_status = voicemail_detected == true and "yes" or "no" freeswitch.consoleLog("info", "Was VM detected? " .. vm_status .. "\n"); if(voicemail_detected) then return "break"; end freeswitch.consoleLog("notice", "Played the message at least once and checked for VM, we should be exiting the loop.\n") end if (voicemail_detected) then freeswitch.consoleLog("info", "Playback for voicemail.\n"); session:execute("avmd","stop"); playbackMessage(5000); end freeswitch.consoleLog("info", "All finished, hanging up the session.\n"); session:hangup(); Any help would be greatly appreciated. Thank you, Michael

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VM to email error
by Spencer Thomason 23 Feb '11

23 Feb '11

Hello all, I keep seeing an error in the logs similar to the following when a VM is emailed. Everything seems to be working just fine so I'm trying to decipher the log entry. 2011-02-21 19:57:48.288269 [ERR] switch_utils.c:699 Unable to execute command: /bin/cat /tmp/mail.12983398676f47 | /usr/sbin/sendmail -f fromuser(a)fromdomain.com -t touser(a)todomain.com /bin/cat works as does /usr/sbin/sendmail. This is on Centos 5.5 w/ Postfix instead of sendmail and /var/log/ maillog clearly shows the message being delivered. Do I have something misconfigured? Thanks, Spencer

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Re: [Freeswitch-users] risky hangup
by Madovsky 23 Feb '11

23 Feb '11

apparently I found the problem, there were 2 instances of FS running... btw, is there any trick (other than killall -9 freeswitch before start the fs startup script) to avoid to run 2 instances of FS ? ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Tuesday, February 22, 2011 7:06 PM Subject: Re: [Freeswitch-users] risky hangup Example: I make an external call without credits so the call is transferred (by nibblebill) to a "nofunds" extension with plaback wav file ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Tuesday, February 22, 2011 6:42 PM Subject: Re: [Freeswitch-users] risky hangup please give some context... how are you testing this? What is calling what, etc.? -MC On Tue, Feb 22, 2011 at 3:26 PM, Madovsky <infos(a)madovsky.org> wrote: on last git I noticed that every hangup after TTS is made with a dfferent delay. sometimes it cuts the voice, sometimes it cuts suddenly before the voice starts and sometimes it cuts the voice 2/3 seconds before. after 10 tests all hangups are done with a different delay... please advice thanks ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Easy IVR?
by EdPimentl 23 Feb '11

23 Feb '11

Is this not a good example of how easy IVR design/creation should be? http://asymmetrical-view.com/2011/02/20/twilio-in-ten-minutes.html

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newbie question: how do you set the folder that freeswitch uses for sounds?
by Edward de Jong 23 Feb '11

23 Feb '11

I see that in the default config, freeswitch uses sounds from the en/us/callie folder. so when you ask for playback of hello.wav, it goes to that folder by default. how does one change the folder used by freeswitch? i want to let the user pick between different voices, which of the many config files sets up which voice to use? and what if I want to have better voice on my user's phone? can I go to 16000 sample rate? if the user has HD capability on their phone, shouldn't I be using better voice samples? and how can I choose the voice folder depending on the phone connected to the PBX at that moment? In other words, if the user has an HD phone can I connect with them at the better codec? obviously if we connect to some other lesser device the dreaded transcoding will occur, but i would think for internal use, for example when the user is navigating menus, they would appreciate a better voice. Or should i just use all waveforms at 16000 and let it drop them down... thanks in advance.

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