- FreeSWITCH-users - FreeSWITCH Mailing Lists

P-Asserted-Identity: "Outbound Call" on internal calls
by Erik Dekkers 04 Nov '11

04 Nov '11

Hey list, Recently (at least since two days) freeswitch is sending UPDATE messages with P-Asserted-Identity: "Outbound Call" <sip:NUMBER@DOMAIN> after a call is established on the callers phone. It looks like it’s not updating the callee’s name with this UPDATE message since the callers name is displayed correctly. This is the dialplan I’m using for internal call routing: <extension name="Local_Extension_SIP"> <condition field="destination_number" expression="^([1-4]\d{2})$"> <action application="bridge" data="user/$1"/> </condition> </extension> Full debug + siptrace: http://pastebin.freeswitch.org/17568 If there’s anything I can test please let me know Regards, Erik Dekkers (wvds-nl) [cid:image78e1b4.JPG@fcaf16f0.48a24e65] ABC Westland 555 Tel: +31 174 245 727 P.O. Box 90 Fax: +31 174 248 739 2685 ZH Poeldijk Mob: +31 624 423 009 www.wvds.nl<http://www.wvds.nl/> erik.dekkers(a)wvds.nl<mailto:erik.dekkers@wvds.nl> ________________________________ Bezoek ons op de Horti Fair / Visit us at the Horti Fair 1-4 november 2011 in Amsterdam, stand 11.0302. [cid:image8e900c.JPG@7fc3c225.4aa01c61]<https://form.n200.com/?s=1dvsast0dsqul&c=2quptgqmwol3s&ac=NTWO000229FIK> ________________________________ Wilk van der Sande is gecertificeerd Philips Led Horti Partner. DISCLAIMER Op alle offertes van, opdrachten aan en overeenkomsten met Bosch Inveka B.V. en Wilk van der Sande B.V. zijn de AVAG-verkoopvoorwaarden van toepassing All our quotations, all orders placed and all contracts concluded with Bosch Inveka B.V. and Wilk van der Sande B.V. are subject to the AVAG-Conditions.

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How to check if user is behind NAT in a programmatic way?
by Mateusz Bartczak 04 Nov '11

04 Nov '11

Hello I would like to allow direct p2p media (bypassing FS) between registered SIP users, but only if one of them is not behind NAT. I can't force bypass_media=true for all calls, because many of users are behind NAT and this will break communication. I only want to bypass_media for users where it's possible. Is there any way of checking this in a programmatic way? I've checked call variables, SIP registration info from database but I can't find any related variable. Any suggestions?

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Newbie
by Jacob.E.Miles＠L-3Com.com 04 Nov '11

04 Nov '11

I am new to FreeSWITCH, testing to make sure it has all the functionality we need, so I have a few questions I would like to ask. I am coming from using Asterisk PBX and some Cisco Call Manager. As you see my question below you will realize we are currently using SCCP (skinny) based phones, I am trying to talk out customer into moving to SIP phones but no luck yet. 1) Is there good SCCP (skinny) support? 2) If so, which phones (79XX)? 3) Is there Asterisk AMI type messaging to control SCCP phones? Jacob Miles Software Engineer III L-3 Communications - Integrated Systems Greenville Jacob.E.Miles(a)L-3Com.com 903.457.4422

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Bad Contact Header
by Dip Mehta 04 Nov '11

04 Nov '11

Hi , I am trying to register x-lite 4.1 on freeswitch. And i am using the default extensions : 1003 and password 1234 However i am unable to get this registered and get the following error on the console. Output of nano /usr/local/freeswitch/conf/directory/default/1003.xml <include> <user id="1003"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1003"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="*1003*"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="*Extension 1003*"/> <variable name="effective_caller_id_number" value="1003"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> recv 572 bytes from udp/[122.176.XX.XX:]:13334 at 16:18:23.774556: ------------------------------------------------------------------------ REGISTER sip:199.167.XX.XX: SIP/2.0 Via: SIP/2.0/UDP 122.176.144.226:13334 ;branch=z9hG4bK-d8754z-9ba4ee0f5e6ae5a7-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1003@122.176.XX.XX:13334;rinstance=cd614363113a396f> To: "Extension 1003"<sip:1003@199.167..XX.XX> From: "Extension 1003"<sip:1003@199.167..XX.XX>;tag=77bad5c7 Call-ID: MTRmNzQ3MDUyMTM4MGNhOGUzZjdlNWYyNWRlZDNjNmQ. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0 ------------------------------------------------------------------------ send 715 bytes to udp/[122.176.XX.XX:]:13334 at 16:18:23.775265: ------------------------------------------------------------------------ * SIP/2.0 401 Unauthorized* Via: SIP/2.0/UDP 122.176.144.226:13334 ;branch=z9hG4bK-d8754z-9ba4ee0f5e6ae5a7-1---d8754z-;rport=13334 From: "Extension 1003"<sip:1003@199.167..XX.XX>;tag=77bad5c7 To: "Extension 1003" <sip:1003@199.167..XX.XX>;tag=gpjScQ0eXXj6K Call-ID: MTRmNzQ3MDUyMTM4MGNhOGUzZjdlNWYyNWRlZDNjNmQ. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7b10506 2011-10-26 08-45-04 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="199.167.145.102", nonce="740c2df0-0637-11e1-9e6e-6b39c14e0024", algorithm=MD5, qop="auth" Content-Length: 0 send 340 bytes to udp/[122.176.144.226]:13293 at 16:10:09.247665: ------------------------------------------------------------------------ * SIP/2.0 400 Bad Contact Header* Via: SIP/2.0/UDP 122.176.XX.XX:;branch=z9hG4bK-d8754z-7c90ddcd2100ff77-1---d8754z-;rport=13293 From: "1003" <sip:1003@199.167.XX.XX>;tag=d09ca84c To: "1003" <sip:1003@199.167.XX.XX>;tag=DU6e75D452FeH Call-ID: YjM3NWZiZjdkMzMwNTJhMjZkNmRhMWFkMGVkZGM1ZWI. CSeq: 4 REGISTER Content-Length: Errors suggest this to be authentication issue. However i am not able to resolve this. Please advice. Regards Dip

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Refer-To does not allow to add 'to-tag' and 'from-tag'
by Mariusz Czulada 04 Nov '11

04 Nov '11

Hi All, I have a problem with my FSW instance (version 1.0.7 on RHER 5.6 64bit). Freeswitch does not allow to pass 'to-tag' or 'from-tag' in addition to "Replaces=" while deflecting a call. This action works: <action application="deflect" data="sip:conf_t0_2@10.79.21.108?Replaces={sip_call_id}" /> The session I try to deflect is set from Broadworks 16 via ALU IMS core (ISC 5450). Our supplier claims that to successfully handle REFER from FSW, Refer-To header should include also 'to-tag' and 'from-tag'. Example I got looks like this: Refer-To: sip:conf_t0_2@10.79.21.108?Replaces=1387BD211B831526@172.16.10.114;to-tag=1728507150-1263982355499;from-tag=xUDOxkDMyYD But when I set action to: <action application="deflect" data="sip:conf_t0_2@10.79.21.108?Replaces=${sip_call_id};to-tag=${sip_to_tag};from-tag=${sip_from_tag}" /> in cosole log I can find: EXECUTE sofia/internal/+48399500015(a)neofon.tp.pl deflect(sip:conf_t0_2@10.79.21.108?Replaces=BW131151222041111-105561828@10.8.80.100;from-tag=139928099-1320408711222-) and FSW freezes until other side issues BYE. Is this a problem on my side (like different action is needed or some vars should be set)? Or is it related to FSW/mod_sofia (source)? TIA for your help or suggesions, Mariusz

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QoS
by Hynek Cihlar 04 Nov '11

04 Nov '11

Hi all, here's an excellent article on Linux QoS http://wiki.linuxwall.info/doku.php/en:ressources:dossiers:networking:traff… . Hynek

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Help with Using Mod_AVMD
by Dave 04 Nov '11

04 Nov '11

Hello, I am creating an external application that uses FreeSWITCH ESL to initiate an outbound call and then run a LUA script. Within the LUA script I am executing AVMD. I want to see if its a human or voicemail on the other end. The outbound call is made and the LUA script executes. Currently I would just like to see a consoleLog entry made by the onInput function in the LUA script, but that function does not seem to be called. AVMD starts the file is streamed then AVMD stops. As shown in the log. I think I'm missing something. Dave Goodwin Pertinent FreeSWITCH Log Info: ------------------------------------------------------------------------------------------------------- EXECUTE sofia/internal/7202126254(a)xxx.xx.xxx.xx avmd(start) 2011-11-03 11:05:27.291678 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/7202126254(a)xxx.xx.xxx.xx 2011-11-03 11:05:28.311249 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16@8000hz 1 channels 20ms 2011-11-03 11:05:28.350313 [DEBUG] switch_rtp.c:3082 Correct ip/port confirmed. 2011-11-03 11:05:40.235535 [DEBUG] switch_ivr_play_say.c:1649 done playing file EXECUTE sofia/internal/7202126254(a)xxx.xx.xxx.xx avmd(stop) 2011-11-03 11:05:40.237488 [DEBUG] switch_core_media_bug.c:467 Removing BUG from sofia/internal/7202126254(a)xxx.xx.xxx.xx Lua Script: ------------------------------------------------------------------------------------------------------- function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '1' and human_detected == false then human_detected = true freeswitch.consoleLog("INFO","Human Detected\n") return "break" end if type == "event" and voicemail_detected == false then voicemail_detected = true freeswitch.consoleLog("INFO","Voicemail Detected\n") return "break" else freeswitch.consoleLog("INFO","Nothing Happened\n") return "break" end end if session:ready() then --- the call has been answered. session:setInputCallback("onInput") session:sleep(1000) session:execute("avmd","start") session:sleep(1000) session:streamFile("ivr/" .. wav_file) session:execute("avmd","stop") session:sleep(1000) fileName = "Completed " .. tostring(call_from_num) .. os.time() .. tostring(call_to_num) freeswitch.consoleLog("INFO","FileName = " .. fileName .. "\n") local myfile, ErrStr = io.open("log/outcall/" .. fileName,"w") if myfile then myfile:write(call_from_num .. "|", call_to_num .. "|" , os.date()) myfile:flush() myfile:close() else freeswitch.consoleLog("INFO","Error writing File = '" .. ErrStr .. "'\n") end --if end --if

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Audio clipping in mod_conference
by Roland Hänel 03 Nov '11

03 Nov '11

Hi, During thorough testing of our conference solution which is based on FreeSwitch / mod_conference, we noticed some 'audio clipping' during the conference. Typically, if a conference member didn't have audio for some time and then begins to speak, the first syllable will be clipped and cannot be heard by the other conference participants. We ruled out a number of possible causes, and finally discovered that the energy level detection algorith seems to be the source of the problem. Setting the energy level for all participants to "0" heals the problem (of course now the noise might be higher since every participant's noise contributes to the total noise), but this setting is not very useful since the "talking detection" seems to rely on the same level, i.e. with energy "0", all members are marked talking all the time (which effectively renders the "talking" display useless). Is there any chance to tune the energy level detection time (!), or to de-couple talking detection an energy level detection in FreeSwitch? Best regards, Roland

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FreeSWITCH - Polycom and DNS SRV
by Lloyd Aloysius 03 Nov '11

03 Nov '11

Hi , I am testing the Polycom Phones [IP335,IP450 and IP6000] and DNS SRV records. All my DNS SRV records and other phone models [Cisico , Asstra, Yealink] working without any issue. But for Polycom Phones .. I need to specify the "Outbound Proxy" as IP address otherwise phone never register. * voIpProt.SIP.outboundProxy.address="IP ADDRESS"* I could not find why we need to specify "Outbound Proxy? How to avoid using IP address? Does Polycom phones fully support DNS SRV records? Any help is appreciated. Thanks and regards, Lloyd

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securing sip channel
by Tamer Higazi 03 Nov '11

03 Nov '11

Hi! I am interested to know if there is a way to secure the registration against account hacking. for example, I would create an SIP Account server side, there would be the possibility that only users with the static IP-Address could login to the assigned account. Is there one other way like kerberos ticket authentication or public secret key with ipsec?! For any advise I would thank you Tamer

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