- FreeSWITCH-users - FreeSWITCH Mailing Lists

Users LANl - Users WAN
by Andrea 29 Jan '09

29 Jan '09

Hi, i came from asterisk and don't understand some things. I have to configure the internal LAN users in directory/default I have to configure the external WAN users in sip_profiles/external (all softphones are behind a router - NAT) I have to configure the gateway (Voip Providers in sip_profiles/internal) it's right? Thank you in advance - Andrea -

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Can't dial over openzap
by "Franziska Röhler" 29 Jan '09

29 Jan '09

Hello, I can't make outbound calls with openzap. I have try to make outbound calls over asterisk and it's work. So I think that the zaptel konfiguration is okay. Or I'm wrong? What's the mistake? I get this error on the console: 2009-01-27 15:32:45 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 111->017XXXX@default 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX [2b330882-1286-4b8b-bfe6-825cee3c8268] 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING] freeswitch.log: 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:339 tech_init() Set codec PCMA 20ms 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:931 channel_outgoing_channel() Connect outbound channel OpenZAP/1:1/017XXXX 2009-01-27 15:32:45 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel OpenZAP/1:1/017XXXX [2b330882-1286-4b8b-bfe6-825cee3c8268] 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:940 channel_outgoing_channel() OpenZAP/1:1/017XXXX State Change CS_NEW -> CS_INIT 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:56 isdn_outgoing_call() Changing state on 1:1 from DOWN to DIALING 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_INIT 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:363 channel_on_init() OpenZAP/1:1/017XXXX State Change CS_INIT -> CS_ROUTING 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:415 switch_core_session_run() (OpenZAP/1:1/017XXXX) State INIT going to sleep 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_ROUTING 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING 2009-01-27 15:32:45 [DEBUG] mod_openzap.c:384 channel_on_routing() OpenZAP/1:1/017XXXX CHANNEL ROUTING 2009-01-27 15:32:45 [DEBUG] switch_ivr_originate.c:57 originate_on_routing() OpenZAP/1:1/017XXXX State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-01-27 15:32:45 [DEBUG] switch_core_session.c:722 switch_core_session_signal_state_change() Kill OpenZAP/1:1/017XXXX [BREAK] 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:420 switch_core_session_run() (OpenZAP/1:1/017XXXX) State ROUTING going to sleep 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() OpenZAP/1:1/017XXXX Running State Change CS_CONSUME_MEDIA 2009-01-27 15:32:45 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (OpenZAP/1:1/017XXXX) State CONSUME_MEDIA 2009-01-27 15:32:45 [ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING] 2009-01-27 15:32:45 [DEBUG] zap_isdn.c:957 q931_rx_32() WRITE 55 -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a

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ATA-answering machine question/recommendation
by jonathan augenstine 29 Jan '09

29 Jan '09

I have an application that requires answering machine detection. I have not been able to locate any documentation indicating that there is explicit support for answering machine detection. I have received recommendations on call flows that would include DTMF entry by the called party to detect by implication answering machines, however, I need an explicit methodology. My question is, does anyone have any experience with ATAs that might have this capability. I am interested in any solution that might even include Avaya, Cisco, or other hardware device interfaced with Freeswitch that would provide an explicit answering machine detection capability. Jonathan

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is there an event for SIP MESSAGE?
by Cavalera Claudio Luigi 29 Jan '09

29 Jan '09

Hello freeswitchers, I'm experimenting with sip clients registered to fs and Instant Messaging. I've seen SIP Messages are properly routed by the sofia sip stack in fs. However it seems no event is ever generated on the event socket. Would it be possible to make fs reporting the SIP Messaging? :-) Best Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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Possible to mock out _freeswitch.py?
by Brian Deacon 29 Jan '09

29 Jan '09

Greetings, Actually any advice on how to test or diagnose my mod_python code would be appreciated. For the most part I've been trying to keep as much of my code as possible separated from the actual interaction with freeswitch so that it's separately testable. But I will occasionally run into: [ERR] mod_python.c:121 eval_some_python() Error importing module This latest is, I think some missing modules in site_packages, but it'd be nice to get some output hinting at that. So nice would be able to have a mocked out _freeswitch so the rest of my unit test stuff could run. Or failing that, is there a switch I can flip to get some of the python errors output to the log? TIA, Brian

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Hello!
by J. G. 28 Jan '09

28 Jan '09

FS users, Just joined the growing list of FS users and found this list. I spend a lot of time in Asterisk (own a 48 seat call center that uses an Asterisk distro as our main switch, etc) and am playing with my first integration between FS and Asterisk, so far so good - looking forward to Anders' release of Phonebooth for the masses. Anyway, glad to be on the list! J. -- ----- Jason Gehman General Manager North Voice Communications www.NorthVC.com

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portaudio error
by e schmidbauer 28 Jan '09

28 Jan '09

hey guys im getting an error when trying to use a USB mic with portaudio. wondering if anyone could help me solve this problem. im using ubuntu intrepid studio. portaudio is loaded correctly and sees the all the devices. when i do a "pa call" using the built in sound card, i am able to connect to the call. but when i set the "pa indev #7" (#7 is the USB mic) and then do a "pa call" i get the following error >pa call 4800 2009-01-28 12:33:09 [NOTICE] switch_channel.c:566 switch_channel_set_name() New Channel portaudio/4800 [bae25c96-ed61-11dd-a1c3-d19a8f52ba15] 2009-01-28 12:33:09 [ERR] mod_portaudio.c:1350 engage_device() Error opening audio device retrying 2009-01-28 12:33:11 [ERR] mod_portaudio.c:1359 engage_device() Can't open audio device 2009-01-28 12:33:11 [NOTICE] mod_portaudio.c:1739 place_call() Close Channel portaudio/4800 [CS_NEW] API CALL [pa(call 4800)] output: FAIL:Device Error! here is the output from "pa devlist" > pa devlist API CALL [pa(devlist)] output: 0;/dev/dsp;16;16;r,o 1;/dev/dsp1;16;0; 2;Intel ICH6: Intel ICH6 (hw:0,0);2;2; 3;Intel ICH6: Intel ICH6 - MIC ADC (hw:0,1);2;0; 4;Intel ICH6: Intel ICH6 - MIC2 ADC (hw:0,2);2;0; 5;Intel ICH6: Intel ICH6 - ADC2 (hw:0,3);2;0; 6;Intel ICH6: Intel ICH6 - IEC958 (hw:0,4);0;2; 7;Samson C01U : USB Audio (hw:1,0);2;0;i 8;front;0;2; 9;default;128;128; 10;dmix;0;2; any input would be much appreciated. thanks again for developing such a fine piece of software!

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DTMF with Early Media Disabled
by Klaus Teller 28 Jan '09

28 Jan '09

Hi, My settings does not allow me to test the following right now. So I'm wondering if somebody knowledgeable could help me answer the following question. I do know that if i call Freeswitch, i can use Javascript to read DTMF even without answering the call. My question is can i do this even if early media is disabled on the inbound call? Thanks, Klaus. -- NUR NOCH BIS 31.01.! GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 EURO/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a

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Sending SIP calls via outbound proxy
by Erick Johnson 28 Jan '09

28 Jan '09

i forgot to give you the pastebin URL http://pastebin.freeswitch.org/6379 > > I'm running latest trunk - Revision: 10682 > > I've been doing an ngrep on my external freeswitch SIP port and FS > is not sending any SIP packets anywhere when I run the following command. > Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are > logged. > > originate > 'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' > &echo() > > Also, just to be clear, when I remove ";fs_path=..." from the command > above a call > is set up normally to erick at ejjohnson.org and the SIP packets are > logged > to console. > > Thanks guys. > > > What SVN rev are you running? Also, could you do a SIP trace? > > TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch > > Pastebin the output of that and we'll take it from there. > > -MC > > > > On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson > > <erick at junctionnetworks.com> wrote: > > > Both: > > > > > > originate sofia/external/'erick at > > ejjohnson.org;fs_path=proxybeta.foo.net' > > > &echo() > > > originate sofia/external/erick at > > ejjohnson.org;fs_path=proxybeta.foo.net > > > &echo() > > > > > > produce the exact same result & log > > > > > > :( > > > > > >> * I think you need to '' the sofia uri /b > > > > > > > > -- Erick Johnson sip/email erick(a)junctionnetworks.com 1-800-801-3381 x7006 Software Engineer Junction Networks

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Another way to set CallerID?
by Ron McCarthy 28 Jan '09

28 Jan '09

Hi, I am having a weird issue with setting the callerID number for outbound calls, I have this: <action application="set" data="effective_caller_id_number=17025551234"/> <action application="bridge" data="sofia/gateway/carrierX/17025559988"/> Ive set the callerID correcly on a Asterisk box and the carrier sees it and passed it correct, but for some reason any calls from Freeswitch won't work, it either shows up as private or another number, so they are not seeing the CID for whatever reason. This method works for other carrirers just fine, but not them. From what I read this sets the from in the SIP message, maybe for some reason that is goofing it up. Any other way to set the callerID in Freeswitch, that's the only way I have found. FS shows it's connecting to a Sonus, does it need to be set different because of some Sonus issue that im not aware of? Any help would be great. Thanks!

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