- FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP Trunk with Private Static IP?
by Roly Maz 26 Feb '10

26 Feb '10

Hi Community My Provider provides the following info when they supply a SIP trunk: . A direct connection into their network. i.e. they provide private IPs: . An IP address I must use for my FS box e.g. IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10. 42.0.68 . An IP address for their SIP server 10.42.0.1 I have setup a dual homed FS box (Windows Server 2008, latest FS version) NIC 1 - Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253 NIC 2 - SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10. 42.0.68 Windows complains about multiple gateways - which I ignore? I can ping internal addresses and the SIP Server When I fire up FS, I can register Xlite phones on my LAN. I can dial and hear the test IVR (5000) This means my Internal SIP Profile is ok. Now, how do i route a call out to the 10.42.01 SIP Server? Creating a gateway doesn't make sense, because I am not supplied a username/password? Any pointers would be most appreciated, I am sure I am missing something really simple. Roland

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Error loading /usr/local/freeswitch/mod/ozmod_sangoma_boost.so
by Nagalenoj H. 26 Feb '10

26 Feb '10

Dear friends, I've installed freeswitch trunk - 16729 and tried to configure with wanpipe for sangoma A102 pri card. Followed the steps given in http://wiki.sangoma.com/wanpipe-freeswitch-install When loading the freeswitch, I've got the following error. 2010-02-23 14:49:58.545726 [INFO] ozmod_wanpipe.c:335 configuring device s2c15 as OpenZAP device 1:30 fd:57 DTMF: software 2010-02-23 14:49:58.546004 [ERR] ozmod_wanpipe.c:436 No valid wanpipe span and channel was specified 2010-02-23 14:49:58.546275 [INFO] zap_io.c:2500 Configured 30 channel(s) 2010-02-23 14:49:58.560724 [ERR] zap_io.c:2560 Error loading /usr/local/freeswitch/mod/ozmod_sangoma_boost.so [/usr/local/freeswitch/mod/ozmod_sangoma_boost.so: cannot open shared object file: No such file or directory] 2010-02-23 14:49:58.561048 [ERR] zap_io.c:2720 can't find 'sangoma_boost' 2010-02-23 14:49:58.561266 [ERR] mod_openzap.c:2458 Error starting OpenZAP span 1 error: 2010-02-23 14:49:58.561712 [NOTICE] switch_loadable_module.c:144 Adding Endpoint 'openzap' Configuration and log files are pasted to pastebin. Kindly someone help me to solve this issue. openzap.conf and openzap.conf.xml http://pastebin.freeswitch.org/12214 freeswitch log http://pastebin.freeswitch.org/12216 smg_pri.conf http://pastebin.freeswitch.org/12217 -- Regards, Nagalenoj H.

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freeswitch to gateway
by srinivasula reddy 26 Feb '10

26 Feb '10

Hi, thank you very munch for reply, this is working fine, when we configure <condition destination_number="^1003" <action application="bridge" data="sofia/gateway/gatewayname/$1"/> but in my scenario, i dont want to route call based on extensions( eg, here 1003) routing, i just want to route the calls when the destination host uri is defferan from local domain, example: INVITE packet from registered extension to sipserver like this. From: 1000@gw.proxy.com:5060 To : 1003@gateway.com:5060 here from uri and to uri is different. any help Srinivas -- Srinivasula Reddy K

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 44, Issue 234
by srinivasula reddy 26 Feb '10

26 Feb '10

Hi, thank you very munch for reply, this is working fine, when we configure <condition destination_number="^1003" <action application="bridge" data="sofia/gateway/gatewayname/$1"/> but in my scenario, i dont want to route call based on extensions( eg, here 1003) routing, i just want to route the calls when the destination domain is defferan from local domain, example: INVITE packet from registered extension to sipserver like this. From: 1000@gw.proxy.com:5060 To : 1003@gateway.com:5060 here from uri and to uri is different. any help Srinivas On Fri, Feb 26, 2010 at 5:44 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Retrieving voicemail without entering user ID > (extension) > (Joseph Puchalski) > 2. Freeswitch/Openzap dials out on disconnected FXO line > (Robert Hadley) > > > ---------- Forwarded message ---------- > From: Joseph Puchalski <joseph.puchalski(a)personalcyberspace.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Date: Thu, 25 Feb 2010 23:31:18 +0000 > Subject: Re: [Freeswitch-users] Retrieving voicemail without entering user > ID (extension) > > Lars, > > > > Thanks! > > > > Joe > > > > *From:* Lars Zeb [mailto:larclap@yahoo.com] > *Sent:* Wednesday, February 24, 2010 10:09 PM > *To:* freeswitch-users(a)lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Retrieving voicemail without entering > user ID (extension) > > > > Joe, > > > > I used the extension below, but I think that Brian said it was too > insecure. Being a total beginner, I removed the condition. > > > > <extension name="Local_Extension_Abe"> > > <condition field="destination_number" expression="^(101[0-9])$"> > > <action application="set" data="dialed_ext=$1"/> > > <action application="export" data="dialed_ext=$1"/> > > </condition> > > <condition field="destination_number" > expression="^${caller_id_number}$"> > > <action application="set" > data="voicemail_authorized=${sip_authorized}"/> > > <action application="answer"/> > > <action application="sleep" data="1000"/> > > <action application="voicemail" data="check default $${domain} > ${dialed_ext}"/> > > <anti-action application="bind_meta_app" data="1 a a > execute_extension::dx XML features"/> > > <anti-action application="bind_meta_app" data="2 a a > record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> > > <anti-action application="bind_meta_app" data="3 a a > execute_extension::cf XML features"/> > > <anti-action application="set" > data="transfer_ringback=${us-ring}"/> > > <anti-action application="set" data="call_timeout=15"/> > > <anti-action application="set" > data="sip_exclude_contact=${network_addr}"/> > > <anti-action application="set" data="hangup_after_bridge=true"/> > >  > > <anti-action application="set" data="continue_on_fail=true"/> > > <anti-action application="db" > data="insert/call_return/${dialed_ext}/${caller_id_number}"/> > > <anti-action application="db" > data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> > > <anti-action application="bridge" data="user/${dialed_ext}@ > $${domain}"/> > > <anti-action application="answer"/> > > <anti-action application="sleep" data="1000"/> > > <anti-action application="voicemail" data="default $${domain} > ${dialed_ext}"/> > > </condition> > > </extension> > > > > Lars > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Rupa > Schomaker > *Sent:* Wednesday, February 24, 2010 4:01 PM > *To:* freeswitch-users > *Subject:* Re: [Freeswitch-users] Retrieving voicemail without entering > user ID (extension) > > > > Look at the end of: > > > > http://wiki.freeswitch.org/wiki/Mod_voicemail#Check_Voice_Mail > > > > Advisable? With it enabled, I can walk up to anyone's phone and retrieve > their VM w/out authentication. It was removed on purpose due to that reason > as far as I remember. > > On Wed, Feb 24, 2010 at 5:45 PM, Joseph Puchalski < > joseph.puchalski(a)personalcyberspace.com> wrote: > > I’m trying to modify my dialplan so that I can press a single button on my > phone, be connected to voicemail, and enter only a password to gain access. > > > > Currently I use a programmable key to dial 4000. I am prompted for my ID, > and then password. > > > > I’ve poked around “mod voicemail” on the wiki and searched the mailing list > and web, but haven’t found enough info. I have discovered that this behavior > seems to have been available in previous versions of the default dialplan. > > > > Is it still possible? Is it advisable? Was this feature/behavior removed > for security reasons? > > > > I apologize ahead of time if the answer is somewhere in plain sight that I > haven’t looked yet. If so, I’d much appreciate being pointed in the right > direction. > > > > As always, thanks for any help, > > > > Joe P. > > > _______________________________________________ > 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 > > > > > -- > -Rupa > > > ---------- Forwarded message ---------- > From: "Robert Hadley" <robert.hadley(a)teotech.com> > To: <freeswitch-users(a)lists.freeswitch.org> > Date: Thu, 25 Feb 2010 16:13:10 -0800 > Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected > FXO line > > When dialing out, Freeswitch/Openzap is not detecting that an analog FXO > channel is disconnected and tries dialing out on the channel anyway. No > error is reported. The call doesn’t timeout until a minute later. > Shouldn’t Freeswitch/Openzap skip over a disconnected channel to the next > connected channel? > > > > I have configured a Sangoma A200 FXO card as a FXO span. > > > > [span wanpipe FXO] > > name => PSTN Line 1 > > number => 4253491059 > > fxo-channel => 2:3 > > name => PSTN Line 2 > > number => 4253491058 > > fxo-channel => 2:4 > > > > > > The wanpipe driver does detect and report when a CO line is connected or > disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as > reported in the log. > > > > /var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3: > FXO Line is disconnected! > > FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT > [ALARM_TRAP][3:1] STATE [DOWN] > > > > /var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4: > FXO Line is connected! > > FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT > [ALARM_CLEAR][3:2] STATE [DOWN] > > > > > > I have the dialplan configured to use the next available port in the FXO > span (there will be more than 2 channels later). > > > > <extension name="OutgoingFXO"> > > <condition field="destination_number" expression="^7(\d+)$"> > > <action application="bridge" data="openzap/FXO/a/$1"/> > > </condition> > > </extension> > > > > > > Here is a portion of the log when that shows dialing out on a disconnected > analog FXO channel. > > > > EXECUTE sofia/internal/5410@192.168.72.45:5060bridge(openzap/FXO/a/93491045) > > 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms > > 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound > channel OpenZAP/3:1/93491045 > > 2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel > OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7] > > 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269 > (OpenZAP/3:1/93491045) State Change CS_NEW -> CS_INIT > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal > OpenZAP/3:1/93491045 [BREAK] > > 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1 > from DOWN to DIALING > > 2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486 > OpenZAP/3:1/93491045 does not support the proxy feature, disabling. > > 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread > starting. > > 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state > handler on 3:1 for DIALING > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/3:1/93491045) Running State Change CS_INIT > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338 > (OpenZAP/3:1/93491045) State INIT > > 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045) > State Change CS_INIT -> CS_ROUTING > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal > OpenZAP/3:1/93491045 [BREAK] > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338 > (OpenZAP/3:1/93491045) State INIT going to sleep > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/3:1/93491045) Running State Change CS_ROUTING > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341 > (OpenZAP/3:1/93491045) State ROUTING > > 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045 > CHANNEL ROUTING > > 2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66 > (OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal > OpenZAP/3:1/93491045 [BREAK] > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341 > (OpenZAP/3:1/93491045) State ROUTING going to sleep > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360 > (OpenZAP/3:1/93491045) State CONSUME_MEDIA > > 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360 > (OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep > > > > > > Thanks, > > Robert > > _______________________________________________ > 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 > > -- Srinivasula Reddy K

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Retrieving voicemail without entering user ID (extension)
by Joseph Puchalski 25 Feb '10

25 Feb '10

I'm trying to modify my dialplan so that I can press a single button on my phone, be connected to voicemail, and enter only a password to gain access. Currently I use a programmable key to dial 4000. I am prompted for my ID, and then password. I've poked around "mod voicemail" on the wiki and searched the mailing list and web, but haven't found enough info. I have discovered that this behavior seems to have been available in previous versions of the default dialplan. Is it still possible? Is it advisable? Was this feature/behavior removed for security reasons? I apologize ahead of time if the answer is somewhere in plain sight that I haven't looked yet. If so, I'd much appreciate being pointed in the right direction. As always, thanks for any help, Joe P.

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Call for help - adding information to the wiki: SIP ALG's
by Michael Collins 25 Feb '10

25 Feb '10

Hi all, I've just completed a new wiki page: http://wiki.freeswitch.org/wiki/ALG I would like all of you who have dealt with routers with SIP ALG's to submit your input. I would like to see this page have a list of how-to's for all of the popular routers. If we can make it easy for people to disable SIP ALG's then I think we can all save ourselves time and energy answering questions in IRC and the mailing lists. Please by all means add your knowledge here. I started with the 2wire 3800HGV that I got for my ATT Uverse service. If you have knowledge that you like to add to the wiki (on this subject or any other) but are not confident in your wiki editing skills then contact me off list and I will be happy to help you get up to speed. Editing your first wiki page is always the hardest... :) Thanks again for all of your help! By the way, today's community conference call was great. Please plan on attending next week and we'll talk about more great FreeSWITCH stuff. I will have the recording of Rupa discussing mod_limit up on line as soon as I can. Take care, Michael

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freeswitch to gateway
by srinivasula reddy 25 Feb '10

25 Feb '10

Hi Good afternoon everybody, my freeswitch domain name is gw.proxy.com, i have registered 1000 to freeswitch, i have configured a gateway(gateway.com) to my freeswitch, i want to make a call to gateway from 1000 to 1003 registered in gateway.com , how can i make call. Thanks-- Srinivasula Reddy K

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qt framework link broken
by Madovsky 25 Feb '10

25 Feb '10

Just to inform that at the link http://wiki.freeswitch.org/wiki/FSComm#Linux the qt framework link is broken, so as I'm new to this emailist I don't want to correct myself on wiki. Regards Franck

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mod_erlang_event
by Mark Sobkow 25 Feb '10

25 Feb '10

It's become clear that I need to use Erlang event processing to do what I need to do with Freeswitch, but I can't even get the most basic of tasks working yet. (i.e. Answer the call and collect the PIN code from the operator.) The dialplan version of what I'm trying to do is: <extension name=\"OpCallsIn\"> <condition field=\"destination_number\" expression=\"^6100$\"> <action application=\"answer\" /> <action application=\"play_and_get_digits\" data=\"4 4 1 5000 # /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav operator_pin \\d+\" /> <action application=\"log\" data=\"INFO operator_pin is ${operator_pin}\" /> <action application=\"log\" data=\"INFO Queueing operator to fifo\" /> <action application=\"fifo\" data=\"6100@${domain_name} in\" /> </condition> </extension> Attached is the Erlang that's attempting to do the same thing. The Erlang is invoked by the following dialplan fragment: <extension name=\"ErlangCallback\"> <condition field=\"destination_number\" expression=\"^6200$\"> <action application=\"erlang\" data=\"pbx_callback pursuit@testsrv\" /> </condition> </extension> Any suggestions? -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com %% Author: mark %% Created: Feb 23, 2010 %% Description: TODO: Add description to pbx_callback -module(pbx_callback). %% %% Include files %% %% %% Exported Functions %% -export([start/0, run/0, launch/1]). start() -> Pid = spawn( ?MODULE, run, [] ), register( ?MODULE, Pid ), { ok, Pid }. run() -> receive { call, Data } -> { event, [UUID | Rest]} = Data, syslog:debug( "pbx_callback:run() New call received, UUID=~p, Rest=~p~n", [UUID, Rest] ), AnswerResults = pbx:api( eval, "uuid:" ++ UUID ++ " answer" ), syslog:debug( "pbx_callback:run() AnswerResults=~p~n", [AnswerResults] ), GetPinResults = pbx:api( eval, "uuid:" ++ UUID ++ " play_and_get_digits 4 4 1 5000 # /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav operator_pin \\d+" ), syslog:debug( "pbx_callback:run() GetPinResults=~p~n", [GetPinResults] ), GetPinVarResults = pbx:api( uuid_getvar, UUID ++ " operator_pin" ), syslog:debug( "pbx_callback:run() GetPinVarResults=~p~n", [GetPinVarResults] ), run(); {call_event, Data} -> { event, [UUID | Rest]} = Data, Name = proplists:get_value( "Event-Name", Rest ), syslog:debug( "pbx_callback:run() call_event UUID=~p, Name=~p, Rest=~p~n", [UUID, Name, Rest] ), run(); {get_pid, UUID, Ref, Pid} -> NewPid = spawn( ?MODULE, run, [] ), syslog:debug( "pbx_callback:run() Request to spawn new handler process, returning PID ~p~n", [NewPid] ), Pid ! { Ref, NewPid }, run() end. launch( Ref ) -> NewPid = spawn( ?MODULE, run, [] ), syslog:debug( "pbx_callback:launch() Returning new PID ~p~n", [NewPid] ), {Ref, NewPid}.

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"hold" tone when dialing into FS
by Peter Olsson 25 Feb '10

25 Feb '10

I guess this is a really stupid question, but I can't find anything about it in my config files... When dialing into the dialplan, and I just execute "answer" and then "park", I get a ring tone played for me. But I just can't find where this ring tone can be specified, it also seems to play if I execute answer and then sleep. I want to replace this ringtone with another sound, so that's why I'm asking... Regards, Peter Olsson

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