January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Announcement: FreeSWITCH Event Socket Library Supports TCL
by Michael Collins 14 Jan '10

14 Jan '10

Just to let everyone know, we've added TCL support for ESL. More info here: http://www.freeswitch.org/node/228 Let us know if you do anything interesting with it and we'll add it to the wiki. -Michael

1 0

Fwd: Re: playback and play_and_get_digits strange misunderstanding
by François Legal 14 Jan '10

14 Jan '10

I did with FreeSWITCH Version 1.0.5-20100112-0400 (hacked) shall I download another one (I don't have svn). François On Thu, 14 Jan 2010 12:20:31 -0500, Michael Jerris wrote: Could you try this with latest trunk as well for comparison, I just changed the way the sounds base directory is set and I want to make sure I did not mess anything up. Mike On Jan 14, 2010, at 3:23 AM, François Legal wrote: So I could (kind of) solve this by myself. There is in vars.xml the variable ${sound_prefix}. I did set it properly to my french sound path and then it worked. However, for the sake of discussion, I did try this with 1.0.5pre8 and the result was different : with the extension FS was trying to play the file ${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav with the extension FS was trying to load ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/ringing_disabled.wav Now with the latest snapshot, with either one of the 2 mentionned extensions, FS tries to play the file in ${sound_prefix}/filepath/${codec_bit_rate}/filename whereas 1.0.5pre8 did not add the codec_bit_rate in the path but took care of the default_language variable. François On Wed, 13 Jan 2010 16:32:16 +0100, François Legal wrote: Hello, trying to make so dialplan extensions that use the playback and play_and_get_digits applications, but I'm having trouble with the file name specification. The files I want to play are in the french language (fr/fr/julie as configured in lang/fr/fr.xml) My extension is as follows : The channel is using a bit rate of 8000 Hz, so by the set default_language=fr I would expect freeswitch to playback the file at ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav whereas it tries to playback the file at ${FREESWITCH_PATH}/sounds/en/us/callie/misc/8000/ringing_disabled.wav I have the same with play_and_get_digits application. What am I doing wrong ? François _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [1] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Links: ------ [1] mailto:FreeSWITCH-users@lists.freeswitch.org

1 0

multiple leg and multiple rtp
by David Villasmil 14 Jan '10

14 Jan '10

hello is it possible to bridge multiple b-legs and provide all audio (progress) until there is an answer on one channel? thanks guys David

9 41

ISP with SIP trunking for 10k+ ports?
by Dale Trub 14 Jan '10

14 Jan '10

We're looking for an ISP with SIP trunking and DIDs to host our FS servers, and are having a hard time locating an ISP who meet these two requirements a) can handle 10,000+ simultaneous inbound ports b) returns our calls in timely manner We have a six-figure annual budget for this service ... and so far the "big guys" sales teams can't get it together to put together a bid. The ideal partner would share with us the backend telcom revenue in the neighborhood of .5c/min, but we're open to other arrangements. Does anyone have any recommendations for ISPs with SIP trunking and DIDs that they can share? Specific salespeople contact info especially appreciated! Thanks, Dale

2 1

Question about a1-hash
by Mike van Lammeren 14 Jan '10

14 Jan '10

Hello! I have written a Lua script to connect to a database and provide directory information for phones registering with FreeSWITCH. My problem is that I store an MD5 hash of the passwords in the database, so I wish there was a way to get FreeSWITCH to authenticate using the MD5 hash of the password provided by the phone, and not the password itself. According to the wiki<http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide>, it is possible to pass in a parameter called *a1-hash* instead of the username and password. The a1-hash parameter is an MD5 hash of a string comprising the username, domain and password, separated by colons. Unfortunately, I can't generate that string, since I don't have the raw password, just the MD5 hash. I would have my Lua script do the authentication, but cannot because FreeSWITCH doesn't pass the user's password to the script. The best solution I can think of is to enter the MD5 hash of the password in the phone. Does anyone have a better idea? Mike van Lammeren

3 8

Re: [Freeswitch-users] playback and play_and_get_digits strange misunderstanding
by François Legal 14 Jan '10

14 Jan '10

So I could (kind of) solve this by myself. There is in vars.xml the variable ${sound_prefix}. I did set it properly to my french sound path and then it worked. However, for the sake of discussion, I did try this with 1.0.5pre8 and the result was different : with the extension FS was trying to play the file ${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav with the extension FS was trying to load ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/ringing_disabled.wav Now with the latest snapshot, with either one of the 2 mentionned extensions, FS tries to play the file in ${sound_prefix}/filepath/${codec_bit_rate}/filename whereas 1.0.5pre8 did not add the codec_bit_rate in the path but took care of the default_language variable. François On Wed, 13 Jan 2010 16:32:16 +0100, François Legal wrote: Hello, trying to make so dialplan extensions that use the playback and play_and_get_digits applications, but I'm having trouble with the file name specification. The files I want to play are in the french language (fr/fr/julie as configured in lang/fr/fr.xml) My extension is as follows : The channel is using a bit rate of 8000 Hz, so by the set default_language=fr I would expect freeswitch to playback the file at ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav whereas it tries to playback the file at ${FREESWITCH_PATH}/sounds/en/us/callie/misc/8000/ringing_disabled.wav I have the same with play_and_get_digits application. What am I doing wrong ? François

2 1

mod_java is moving to unsupported on Jan 20th.
by Brian West 14 Jan '10

14 Jan '10

If nobody steps up to take over maintenance of mod_java by the 20th of January we'll be moving it to unsupported. If you wish to take over maintenance please contact me off list ASAP. Thanks, Brian West FreeSWITCH

1 0

FS can't direct call to SBC
by Vadim Gribanovsky 14 Jan '10

14 Jan '10

Hi all! Let me ask FS experts and users to help to resolve following problem: ///////////////////////////////////////////////////////////////////////////////////////////////// 1.VOIP test environment: =>local (10.10.10.235/24) FS(v1.0.4) installation + =>local IP-Phone (GXP2020) + =>SBC (acts as stateless proxy) provides secure access (TLS +SRTP) to FS for + =>external NATed IP-PHONE (PJSIPua) as depicted below: PHONE(10.10.10.31) <-udp-> FS(10.10.10.235) <-tcp-> (Local 10.10.10.238)SBC(Public P.U.B.LS)<-tls-> INTERNET<-tls-> NAT(Public P.U.B.LN) <-tls-> PHONE (10.10.10.21) 2.Problem: => local IP-Phone get registration to FS, -OK! => external NATed IP-PHONE get registration to FS via SBC, -OK! => FS try to send NOTIFY request to external NATed IP-PHONE using [sip_contact_host:port] record instead [sip_received_ip:port](SBC) - FAIL! => external NATed IP-PHONE calls local IP-Phone - OK! => local IP-Phone couldn't call external NATed IP-PHONE - FS redirect to voicemail - FAIL! 3. Logs: =>FS CLI API CALL [sofia(status)] output: Name Type Data State ================================================================================================= vfsp_profile profile sip:mod_sofia@10.10.10.235:5090 RUNNING (0) vfsp_gw gateway sip:FS_VFSP@10.10.10.238 NOREG internal profile sip:mod_sofia@10.10.10.235:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) 10.10.10.235 alias internal ALIASED ================================================================================================= 3 profiles 1 alias API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.10.10.235 Ext-RTP-IP 10.10.10.235 SIP-IP 10.10.10.235 Ext-SIP-IP 10.10.10.235 URL sip:mod_sofia@10.10.10.235:5060 BIND-URL sip:mod_sofia@10.10.10.235:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 3 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: db639b76d2884388(a)10.10.10.31 User: 1010(a)10.10.10.235 Contact: "user" <sip:1010@10.10.10.31:5060;transport=udp;user=phone> Agent: Grandstream GXP2020 1.2.1.4 Status: Registered(UDP)(unknown) EXP(2010-01-14 16:00:09) Host: aq IP: 10.10.10.31 Port: 5060 Auth-User: 1010 Auth-Realm: 10.10.10.235 ================================================================================================= API CALL [sofia(status profile vfsp_profile)] output: ================================================================================================= Name vfsp_profile Domain Name N/A DBName sofia_reg_vfsp_profile Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.10.10.235 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 10.10.10.235 Ext-SIP-IP P.U.B.L URL sip:mod_sofia@10.10.10.235:5090 BIND-URL sip:mod_sofia@10.10.10.235:5090 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: 6197eb4f-72fa-428f-8c0e-5c5551d185e2 User: 1007(a)10.10.10.235 Contact: "user" <sip:1007@P.U.B.LN:43459;transport=tcp> Agent: PJSUA v1.0.3/i686-pc-linux-gnu Status: Registered(TCP)(unknown) EXP(2010-01-14 16:04:46) Host: aq IP: 10.10.10.238 Port: 38782 Auth-User: 1007 Auth-Realm: 10.10.10.235 ================================================================================================= => sniffer pkt1. SBC -> FS REGISTER sip:10.10.10.235;transport=tcp SIP/2.0 Record-Route: <sip:10.10.10.238:5060;transport=tcp;lr> Via: SIP/2.0/tcp 10.10.10.238:36707;rport;branch=z9hG4bKPj66cc4b809a6bb1510af258991c1f0587 Via: SIP/2.0/tcp 10.10.10.21:32780;rport=43459;received=P.U.B.L;branch=z9hG4bKPj71ce8307-a505-4477-9c47-348cb947b1be Max-Forwards: 69 From: <sip:1007@10.10.10.235>;tag=a702b4fb-d565-4627-9cff-5fb6bfed2922 To: <sip:1007@10.10.10.235> Call-ID: b618fe07-a1ef-4eee-926c-ff8e8f3c9096 CSeq: 49600 REGISTER User-Agent: PJSUA v1.0.3/i686-pc-linux-gnu Contact: <sip:1007@P.U.B.LN:43459;transport=tcp> Expires: 300 pkt2. FS -> SBC SIP/2.0 200 OK Via: SIP/2.0/TCP 10.10.10.238:36707;rport=36707;branch=z9hG4bKPj66cc4b809a6bb1510af258991c1f0587 Via: SIP/2.0/TCP 10.10.10.21:32780;rport=43459;received=P.U.B.LN;branch=z9hG4bKPj71ce8307-a505-4477-9c47-348cb947b1be From: <sip:1007@10.10.10.235>;tag=a702b4fb-d565-4627-9cff-5fb6bfed2922 To: <sip:1007@10.10.10.235>;tag=Brpemg1NcteDc Call-ID: b618fe07-a1ef-4eee-926c-ff8e8f3c9096 CSeq: 49600 REGISTER Contact: <sip:1007@P.U.B.LN:43459;transport=tcp>;expires=300 Date: Thu, 14 Jan 2010 13:15:37 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported pkt3. FS -> external NATed IP-PHONE (trying to send NOTIFY via default gateway instead SBC!!!) Internet Protocol, Src: 10.10.10.235 (10.10.10.235), Dst: P.U.B.LN (P.U.B.LN) Transmission Control Protocol, Src Port: 52304 (52304), Dst Port: 43459 (43459), Seq: 0, Len: 0 Flags: 0x02 (SYN) 4. default gateway -> FS Internet Protocol, Src: P.U.B.LN (P.U.B.LN), Dst: 10.10.10.235 (10.10.10.235) Transmission Control Protocol, Src Port: 43459 (43459), Dst Port: 52304 (52304), Seq: 1, Ack: 1, Len: 0 Flags: 0x14 (RST, ACK) /////////////////////////////////////////////////////////////////////////////////////////////////////////// The same happened when i use default FS configuration (with external profile). Can somebody push me to the right way? Thanks in advance, Vadim

1 0

ESL Mono/Managed
by Brian West 14 Jan '10

14 Jan '10

I need someone that knows C# to write up some examples using ESL in C# cd libs/esl make managedmod cd managed Look at the perl example single_command.pl require ESL; my $command = shift; my $args = join(" ", @ARGV); my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); my $e = $con->api($command, $args); print $e->getBody(); Need to write the same thing in C# and commit it as an example. Also other examples exist in the perl directory those should be translated also if possible. If you're interested please find me on IRC (bkw_) Thanks, Brina

2 1

How to build FreeSwitch with non-default SSL library?
by Yehavi Bourvine 14 Jan '10

14 Jan '10

Hello, I suspect that the Openssl library used on fedora is a modified one and has some interoperability problems with phones. I would like to build freeswitch with a library from openssl.org which is installed under /usr/local. Is there an easy way to tell "configure" script to use that version instead? Thanks, __Yehavi:

1 1