- FreeSWITCH-users - FreeSWITCH Mailing Lists

Pulse dialing in freeswitch.
by Samat Galimov 21 Mar '12

21 Mar '12

Hello! Is it possible to make outgoing calls to PSTN via pulse dial? My PSTN provider uses old hardware, so they doesnt support tone dial, and I have some small SIP gateways (Linksys SPA3102) which dont support pulse dial. Can I generate tones on the freeswitch side? Thank you.

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Originate using inbound socket connection
by Michael Lutz 21 Mar '12

21 Mar '12

Hi Guys, I am working on a .net ESL host (inbound mode) and connect to the ESL using ESLConnection. Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. It always returns: === Content-Type: api/response Content-Length: 125 Content-Length: 125 -USAGE: <call url> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] === But I am pretty sure my string is right: {origination_caller_id_number=31341.....,origination_caller_id_name=31341......}sofia/external/31634258...(a)xxx.xxx.xxx.xxx I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. My code: ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); if (eslDial.Connected() == ESL_SUCCESS) { // Create a uuid used to identify the b-leg. string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); string cDialString = "{{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + thisDestination; // Send the command var eslEvent = eslDial.Api("originate", cDialString); // Write the result to the console Console.WriteLine(eslEvent.Serialize(string.Empty)); return true; } Thanks for your help!, Regards, Michael Lutz

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BLF/Presence
by Brett Wilson 21 Mar '12

21 Mar '12

Hey guys, Im looking to get BLF working on my GXP2100's. I have not been able to find much info on how to configure it. I have set the phones keys to BLF, so they constantly light up green. But they never change. I have those keys configured for the other extensions, ie. 101, 102, 103 etc. Can someone steer me in the right direction? Thanks. Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson(a)launch3.net AOL IM: Brett.Wilson(a)launch3.net <blocked::http://www.launch3.net/> www.Launch3.net <http://www.launch3telecom.com/> www.Launch3telecom.com *************************** <http://launch3telecom.blogspot.com/> Description: Description: Blogger-logo <http://www.facebook.com/pages/Launch-3-Telecom/165610820178720> Description: Description: FaceBook-Logo <https://twitter.com/launch3telecom> Description: Description: Twitter-Logo

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Setting effective_caller_id_name before putting a user in a callcenter queue has no effect
by Markus Lindenberg 21 Mar '12

21 Mar '12

Hi, my dialplan goes like this: <action application="export" data="effective_caller_id_name=HOTLINE ${caller_id_name}"/> <action application="callcenter" data="hotline@${domain_name}"/> Setting/exporting effective_caller_id_name has no effect when the call is presented to a callback agent. the original callerid and calleridname will be used. for now i worked around the problem like this: <extension name="hotline"> <condition field="destination_number" expression="^42$"> <action application="export" data="effective_caller_id_name=HOTLINE ${caller_id_name}"/> <action application="bridge" data="loopback/hotline_queue"/> </condition> </extension> <extension name="hotline_queue"> <condition field="destination_number" expression="^hotline_queue$"> <action application="callcenter" data="hotline@${domain_name}"/> </condition> </extension> while this works, it just doesn't look right to me and means i have to take additional actions if i want the dialplan to continue after bridging into the queue. To me the obvious solution would be to have mod_callcenter.c (line 2479 "INSERT INTO members...") check for origination/effective caller id variables and use them, if available. I'm already doing this kind of check over and over in some lua code i call before bridging (e.g. for sending xmpp notifications): cidnum = session:getVariable("effective_caller_id_number") if not cidnum then cidnum = session:getVariable ("origination_caller_id_number") end if not cidnum then cidnum = session:getVariable ("caller_id_number") end This doesn't feel entirely right either. Shouldn't there be a command that always gives me the computed callerid after possible manipulations? or should i generally do this stuff in the b leg, where the effective_caller_id_name is available as caller_id_name? Thanks, Markus

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good Fax T.38 Provider
by Anita Hall 21 Mar '12

21 Mar '12

Hi I am looking for good T.38 Providers for both inbound and outbound Fax application. Any recommendation ? (I am using mod_spandsp) regards, Anita

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How can I change the moh_sound
by piyush singhai 21 Mar '12

21 Mar '12

Hello, I want to change the moh sound at run time for conference. can i specify path on the basis of any ivr. --Piyush

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TLS on FS
by Tihomir Culjaga 21 Mar '12

21 Mar '12

hello, im new on TLS setup... and as usual, im having issues configuring a SIP client (Bria for windows) with FS. I guess i configured FS properly, but im not sure about certificates. FS conf: <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/> I created certificates using the commands on the wiki: ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org ./gentls_cert create_server -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org ./gentls_cert create_client -cn Client1 -out Client1 /usr/local/freeswitch/conf/ssl -rw-r----- 1 root root 3029 Mar 20 08:56 agent.pem drwxr-x--- 2 root root 4096 Mar 20 08:56 CA -rw-r----- 1 root root 1046 Mar 20 08:49 cafile.pem -rw-r----- 1 root root 3029 Mar 20 09:45 Client1 /usr/local/freeswitch/conf/ssl/CA -rw-r----- 1 root root 1046 Mar 20 08:49 cacert.pem -rw-r----- 1 root root 17 Mar 20 09:45 cacert.srl -rw-r----- 1 root root 1679 Mar 20 08:49 cakey.pem -rw-r----- 1 root root 579 Mar 20 08:49 config.tpl i deployed and installed cafile.pem on windows machine running Bria softphone. I did the same with Client1. Restarted but all im getting is this error in console: tport_wakeup_pri(0x85b9590): events IN tport_alloc_secondary(0x85b9590): new secondary tport 0x85ef610 tport_tls_accept(0x85ef610): new connection from tls/ 85.114.34.202:61030/sips tls_connect(0x85ef610): events NEGOTIATING tls_connect(0x85ef610): events NEGOTIATING tls_connect(0x85ef610): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport_close(0x85ef610): tls/85.114.34.202:61030/sips please, can anyone help ? this is a portion showing how FS loads mod_sofia: su_port_create(0x857c060): epoll_create() => 0: OK su_socket_port_init(0x857c060, 0xd9b400) called su_pthread_port_init(0x857c060, 0xd9b400) called su_port_create(0x85b1840): epoll_create() => 0: OK su_socket_port_init(0x85b1840, 0xd9b400) called su_pthread_port_init(0x85b1840, 0xd9b400) called nua: nua_create: entering su_port_create(0x85b3920): epoll_create() => 0: OK su_socket_port_init(0x85b3920, 0xd9b400) called su_pthread_port_init(0x85b3920, 0xd9b400) called nua: nua_stack_init: entering nua: nua_stack_set_params: entering soa_create("default", 0x85b0eb8, 0x85b0f70) called soa_set_params(static::0x85ae410, ...) called soa_set_params(static::0x85ae410, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nua: nua_create: entering su_port_create(0x85b2368): epoll_create() => 0: OK su_socket_port_init(0x85b2368, 0xd9b400) called su_pthread_port_init(0x85b2368, 0xd9b400) called nua: nua_stack_init: entering nua: nua_stack_set_params: entering soa_create("default", 0x85b2198, 0x85b6440) called soa_set_params(static::0x85b6938, ...) called soa_set_params(static::0x85b6938, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nta_agent_create: initialized random identifiers nta_agent_create: initialized timer nta_agent_create: initialized random identifiers nta_agent_create: initialized timer nta_agent_create: initialized resolver nta_agent_create: initialized resolver tport_create(): 0x85aefe8 tport_create(): 0xb780f688 nta: master transport created nta: master transport created tport_bind_server(0x85aefe8) to */85.114.35.241:5060/sip tport_bind_server(0xb780f688) to */85.114.35.241:5080/sip tport_bind_server(0xb780f688): calling tport_listen for udp tport_bind_server(0x85aefe8): calling tport_listen for udp tport_alloc_primary(0x85aefe8): new primary tport 0x85af2d0 tport_alloc_primary(0xb780f688): new primary tport 0xb780add0 tport_listen(0xb780add0): listening at udp/85.114.35.241:5080/sip tport_listen(0x85af2d0): listening at udp/85.114.35.241:5060/sip tport_bind_server(0xb780f688): calling tport_listen for tcp tport_bind_server(0x85aefe8): calling tport_listen for tcp tport_alloc_primary(0xb780f688): new primary tport 0xb7811af0 tport_alloc_primary(0x85aefe8): new primary tport 0x85af778 tport_listen(0x85af778): listening at tcp/85.114.35.241:5060/sip tport_listen(0xb7811af0): listening at tcp/85.114.35.241:5080/sip nta: bound to (85.114.35.241:5080;transport=*) nta: bound to (85.114.35.241:5060;transport=*) nta: agent_init_via: SIP/2.0/udp 85.114.35.241:5080 (sip) nta: agent_init_via: SIP/2.0/udp 85.114.35.241 (sip) nta: agent_init_via: SIP/2.0/tcp 85.114.35.241 (sip) nta: agent_init_via: SIP/2.0/tcp 85.114.35.241:5080 (sip) nta: Via fields initialized nta: Via fields initialized nta: Contact header created nta: Contact header created tport_bind_server(0x85aefe8) to tls/85.114.35.241:5061/sips tport_bind_server(0xb780f688) to tls/85.114.35.241:5081/sips tport_bind_server(0xb780f688): calling tport_listen for tls tport_bind_server(0x85aefe8): calling tport_listen for tls tport_alloc_primary(0xb780f688): new primary tport 0xb780e378 tport_alloc_primary(0x85aefe8): new primary tport 0x85b9590 tport_tls_init_master(0x85b9590): tls key = /usr/local/freeswitch/conf/ssl/agent.pem tport_tls_init_master(0xb780e378): tls key = /usr/local/freeswitch/conf/ssl/agent.pem tport_tls_init_master(0xb780e378): tls context initialized for [85.114.35.241]:5081 tport_tls_init_master(0x85b9590): tls context initialized for [85.114.35.241]:5061 tport_listen(0x85b9590): listening at tls/85.114.35.241:5061/sips tport_listen(0xb780e378): listening at tls/85.114.35.241:5081/sips nta: bound to (85.114.35.241:5061;transport=tls) nta: bound to (85.114.35.241:5081;transport=tls) nta: agent_init_via: SIP/2.0/udp 85.114.35.241 (sip) nta: agent_init_via: SIP/2.0/udp 85.114.35.241:5080 (sip) nta: agent_init_via: SIP/2.0/tcp 85.114.35.241 (sip) nta: agent_init_via: SIP/2.0/tcp 85.114.35.241:5080 (sip) nta: agent_init_via: SIP/2.0/tls 85.114.35.241 (sips) nta: agent_init_via: SIP/2.0/tls 85.114.35.241:5081 (sips) nta: Via fields initialized nta: Via fields initialized nta: Contact header created nta: Contact header created nua_register: Adding contact URL '85.114.35.241' to list. nua_register: Adding contact URL '85.114.35.241' to list. nua_register: Adding contact URL '85.114.35.241' to list. nua_register: Adding contact URL '85.114.35.241' to list. nua: nua_set_params: entering nua: nua_set_params: entering nua((nil)): sent signal r_set_params nua((nil)): sent signal r_set_params nua: nua_stack_set_params: entering soa_set_params(static::0x85b6938, ...) called 2012-03-20 10:57:48.859351 [NOTICE] sofia_reg.c:2969 Added gateway ' example.com' to profile 'external' nua: nua_stack_set_params: entering soa_set_params(static::0x85ae410, ...) called 2012-03-20 10:57:48.859745 [NOTICE] sofia.c:2710 Adding Alias [85.114.35.241] for profile [internal] tport_wakeup_pri(0x85b9590): events IN tport_alloc_secondary(0x85b9590): new secondary tport 0x85bae68 tport_tls_accept(0x85bae68): new connection from tls/ 85.114.34.202:60916/sips tls_connect(0x85bae68): events NEGOTIATING tls_connect(0x85bae68): events NEGOTIATING tls_connect(0x85bae68): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport_close(0x85bae68): tls/85.114.34.202:60916/sips nua: nua_application_event: entering nua: nua_application_event: entering 2012-03-20 10:57:49.959654 [CONSOLE] sofia.c:1214 MSG Thread Started nua: nua_handle_magic: entering nua: nua_handle_magic: entering 2012-03-20 10:57:50.360893 [CONSOLE] switch_loadable_module.c:1299 Successfully Loaded [mod_sofia]

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Re: [Freeswitch-users] Advice how to start with H.323
by Peter Olsson 20 Mar '12

20 Mar '12

Thanks! /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Peter Steinbach Skickat: den 20 mars 2012 15:39 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Advice how to start with H.323 Wiki is updated accordingly. Best regards Peter Am 20.03.2012 13:32, schrieb Peter Olsson: > It would be very appreciated if you could take some time and update with some more information on the wiki - so others could benefit from this as well.. > > /Peter > > > -----Ursprungligt meddelande----- > Från: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Peter > Steinbach > Skickat: den 20 mars 2012 13:08 > Till: FreeSWITCH Users Help > Ämne: Re: [Freeswitch-users] Advice how to start with H.323 > > Thanks to your help, I am now a step further with 1.22.0 and ptlib-2.8.2: > > mod_h323 has compiled successfully. > After relinking libh323_linux_x86_64_.so.1.22.0 the mod_h323 module finally loaded. > Thanks to all, who helped on this issue! > > Do you know of any gatekeeper avaliable for testing, where I could test connectivity and 2-way-audio? > I tried with http://www.voxgratia.org/ but this gateway seems to be down. > > Best regards > Peter > > > > > Am 19.03.2012 14:33, schrieb Peter Olsson: >> Yes, I've used those recommendations - so 1.22.0 is probably a better choice. >> >> /Peter >> >> >> -----Ursprungligt meddelande----- >> Från: freeswitch-users-bounces(a)lists.freeswitch.org >> [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Patrick >> Lists >> Skickat: den 19 mars 2012 14:03 >> Till: FreeSWITCH Users Help >> Ämne: Re: [Freeswitch-users] Advice how to start with H.323 >> >> On 19-03-12 11:53, Peter Steinbach wrote: >>> I am using Ubuntu 10.04. >>> >>> I followed the installation procedures on the wiki page and >>> installed >>> >>> ptlib-2.8.2 + h323plus-trunk >> Not sure if it's still valid but in the past the recommendation was to follow the versions listed at: >> >> http://www.gnugk.org/compiling-gnugk.html >> >> Which in this case means you should use H323Plus 1.22.0 and not trunk. >> >> Regards, >> Patrick >> >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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-us >> e >> rs >> http://www.freeswitch.org >> >> >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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-us >> e >> rs >> http://www.freeswitch.org >> > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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-use > rs > http://www.freeswitch.org > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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-use > rs > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:4f68950732764931015849!

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TDM400P spans
by Jeremy Johnson 20 Mar '12

20 Mar '12

I'm trying to convert from Asterisk to Freeswitch, and have a question about spans. I read the freetdm configuration example for TDM400 and am wondering what exactly is a span. I have 2 TDM400P cards in my computer, one card has 2 FXS modules for connecting analog phones. The other card has 3 FXO modules for connecting to 3 PSTN trunk lines. # lsdahdi ### Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" 1 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) 2 FXS FXOKS (In use) (EC: OSLEC - INACTIVE) 3 unknown Reserved 4 unknown Reserved ### Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) 5 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) 6 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) 7 FXO FXSKS (In use) (EC: OSLEC - INACTIVE) 8 unknown Reserved Do I configure freetdm as in the example for TDM400 with separate spans for each module like the following? or should I have just two spans as shown by lsdahsi? Thanks. #cat freetdm.conf [general] cpu_monitor => yes cpu_monitoring_interval => 2000 ; monitor usage every 2 seconds cpu_set_alarm_threshold => 90 ; whenever 90% of global CPU usage is reached, trigger the alarm. cpu_reset_alarm_threshold => 80 ; when the CPU usage decreases at 80%, clear the alarm. cpu_alarm_action => reject,warn ; Start rejecting calls when the CPU alarm is triggered and also print warnings. ; ANALOG PHONES [span zt FXS1] fxs-channel => 1 analog-start-type => kewl ;txgain => ;rxgain => ;number => [span zt FXS2] fxs-channel => 2 analog-start-type => kewl ;txgain => ;rxgain => ;number => ; PSTN TRUNK LINES [span zt FXO5] trunk-type => FXO fxo-channel => 5 analog-start-type => kewl ;txgain => ;rxgain => ;number => [span zt FXO6] trunk-type => FXO fxo-channel => 6 analog-start-type => kewl ;txgain => ;rxgain => ;number => [span zt FXO7] trunk-type => FXO fxo-channel => 7 analog-start-type => kewl ;txgain => ;rxgain => ;number =>

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Mod_CallCenter and HA Failover
by R H 20 Mar '12

20 Mar '12

Hey Guys, First I want to say that FreeSwitch has been an amazing platform to work on. Compared to asterisk I have been extremely impressed. In the callcenter.conf.xml I noticed a note that said "  **" There is not much more elaboration on the topic anywhere to be found so I was wondering if there was ever a plan to support HA Failover in callcenter. I recently spent some time working on a successful HA failover setup in hopes to ensure uptime on our servers. Basic call flows work perfectly but CallCenter definitely shows negative results. Any feedback you can give me about plans for the module? Thanks! Ryan PS - If it's welcome, I'm planning on adding the notes of how I got HA Failover working with 2 servers to the FreeSwitch wiki. The documentation there is a bit incomplete. I'll try to do that as soon as I feel the notes are complete.

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