- FreeSWITCH-users - FreeSWITCH Mailing Lists

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

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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

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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

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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

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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

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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

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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:

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Issue with openzap and DTMF
by François Legal 14 Jan '10

14 Jan '10

Hello, yesterday, while trying some new extensions with play_and_get_dtmf, I noticed a strange behaviour of freeswitch. The configuration is as follows : An FXS port is bridged to an FXO (say FXO1) port for calling (through my carrier) another FXO (say FXO2) port. The call incoming on FXO2 reaches some extension which does answer then play_and_get_dtmf (or read). So I dial the digits I want, and for each dialed digit, I get it twice in the resultant variable. Now if I reach the extension by calling in from a cell phone, I get exactly the digits I dialed. This seems to me as a problem in openzap. I just wanted to report. François

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Attend CampKDE Jan 15-18 via Voice over Internet (VOIP), BerkeleyTIP
by giovanni_re 14 Jan '10

14 Jan '10

This weekend is CampKDE in San Diego, California, USA. For those who would like to participate, but can't physically attend, the BerkeleyTIP group will (unofficially, as a public service) provide assistance to everyone who would like to communicate via voice about &/or with the conference, & other KDE community members, globally. [VOIP communication is typically free (no money $ cost) to people with internet access.] Join #berkeleytip on irc.freenode.net, & we will do our best (depending on availability of BTIP people) to help people get on VOIP in order to talk about KDE, or whatever. You are also invited to join the BerkeleyTIP mailing list, & ask any help or other questions there. http://groups.google.com/group/BerkTIPGlobal Here's the announcement there: CampKDE Jan 15-18 SanDiego, VOIP & Ride Share; BerkeleyTIP http://groups.google.com/group/BerkTIPGlobal/t/53cbf00b987b66e0 I'll be announcing this on some KDE related mailing lists. When (if) I get time, I'll locate the CampKDE organizers, & email them telling them BTIP is helping KDE community members with this unofficially, & asking them if they'd like to try to do something officially in arranging global VOIP for the KDE community interested in CampKDE. Here is the VOIP HowTo page for BerkeleyTIP: http://sites.google.com/site/berkeleytip/remote-attendance There are 2 ways to join the VOIP conference: 1) Computer VOIP globally using a sw client & headset & mic, & 2) Dial in with land or cell phone to a phone # in Washington state, USA (free, no $ money cost, if you have free long distance phone service). BerkeleyTIP: http://sites.google.com/site/berkeleytip/ ===== So, join on the #berkeleytip freenode irc channel, or the mailing list, & we'll do our best to help you get on the global conference to discuss KDE via voice communication. :)

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Sangoma PRI installation for FreeSWITCH
by Magesh R 14 Jan '10

14 Jan '10

Dear All, I have installed Sangoma PRI card in machine with following steps, * wget ftp://ftp.sangoma.com/linux/custom/3.5/wanpipe-3.5.8.7.tgz * tar -xvfz wanpipe-3.5.8.7.tgz * cd wanpipe-3.5.8.7 * make openzap * make install * make install_pri I have executed "wanrouter hwprobe" command it prints the following details, 1 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=1 : HWEC=0 : V=36 2 . AFT-A102-SH : SLOT=4 : BUS=3 : IRQ=11 : CPU=A : PORT=2 : HWEC=0 : V=36 Card Cnt: A101-2=1 Next I have executed wancfg_fs script to configure the sangoma for freeswitch. It creates the following configuration files * wanpipe1.conf * wanpipe2.conf * smg_prid.conf * openzap.conf * openzap.conf.xml I have attached those files. I have started the wanrouter and printed the wanrouter status, Wanrouter Status: Device name | Protocol | Station | Status | wanpipe1 | AFT TE1 | N/A | Connected | wanpipe2 | AFT TE1 | N/A | Disconnected | Next I have started the smg_ctrl, but it failed to start. It prints the following things, smg_ctrl start Starting processes... Loading SCTP...OK Starting sangoma_prid...OK sangoma_prid failed to start check /var/log/sangoma_mgd.log for errors Stopping running processes... safe_sangoma not running... sangoma_prid is stopped Removing PID files...done I have checked /var/log/sangoma_mgd.log file. But nothing was there. Could any please tell me where I made mistake? Thanks, Mag

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