- FreeSWITCH-users - FreeSWITCH Mailing Lists

Mobile callback + DISA
by FreeSWITCH-Users 20 Jun '10

20 Jun '10

Hi all, I am new to FS but have been working with * for a while.. I have * setup for callback+DISA used by family for cellular minutes. Have been looking at FS this past month or so as I'd like to setup FS on my router with DD-WRT + FS. Presently I have figured out how to setup disa.js for DISA on my cell BUT I can't seem to find anything out on the web about configuring callback. Has anyone figured out how to setup callback+DISA? Any help is appreciated, links or any ideas would help me over this hurdle. Cheers, S.

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mod_lcr update - read if you use custom_sql
by Rupa Schomaker 20 Jun '10

20 Jun '10

mod_lcr has been updated. commit log: mod_lcr update. MODAPP-340, MODAPP-355 arbitrary b-leg vars, limit support, lcr/ endpoint custom_sql now is field based rather than position based NOTE: custom_sql is incompat with prior version arbitrary b-leg vars enable integration with mod_nibblebill. Wiki has been updated: http://wiki.freeswitch.org/wiki/Mod_lcr This change is backwards compatible EXCEPT if you use custom_sql. If you use custom_sql you must update your sql to use field aliases (SQL : AS). The module no longer uses position in the custom sql for fields. So, if you use custom_sql, reread: http://wiki.freeswitch.org/wiki/Mod_lcr#Custom_SQL Ok, I'm done breaking things for the day. :) -- -Rupa

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continue_on_fail and hangup_after_bridge with transfer
by Vladimir Elizarov 19 Jun '10

19 Jun '10

I'm trying to make a dialplan: <extension name="Long Dial"> <condition field="destination_number" expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^\+([0-9]{11})$)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="failure_causes=TIMEOUT,NO_ROUTE_DESTINATION"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/> <action application="set" data="auto_hunt=true"/> <action application="set" data="real_dialed_number=7777777"/> </condition> <condition field="destination_number" expression="^(79|89)(\d{9})$" break="on-true"> <action application="set" data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/> <action application="bridge" data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/> <action application="transfer" data="to_card"/> </condition> </extension> <extension name="to_card"> <condition field="destination_number" expression="^to_card$"> <action application="bridge" data="{execute_on_answer=send_dtmf\s${dtmf}}sofia/internal/${real_dialed_number}@192.168.50.53:5061"/> </condition> </extension> The logic of his work: if unavailable ilti gateway to the next. if not available all the gateway to go to the extension transfer. A problem arises when user_busy on the phone: ------------------------------------------------------------------------ SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 192.168.50.11:5080;rport;branch=z9hG4bK1X4r00t93KmFg From: "Vladimir Elizarov" <sip:0000xxxx@xx.xx.xx.xx;transport=udp>;tag=yj15KpSQvK6eF To: <sip:8931237xxxx@xx.xx.xx.xx>;tag=2a5549004a431d10ff00001a646aff45 Call-ID: 1317949e-f694-122d-8fb6-00163efcbed2 CSeq: 132374757 INVITE Server: MERA MSIP v.1.0.2 Reason: Q.850;cause=17;text="User busy" Content-Length: 0 instead of making hangup, dialplan is going to transfer. Constant despite continue_on_fail = NORMAL_TEMPORARY_FAILURE, TIMEOUT, NO_ROUTE_DESTINATION" What's the problem? freeswitch 1.0.6 (git 19 june 2010) -- Best regards, Vladimir Elizarov

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Build the mod_unimrcp module
by Thangappan.M 19 Jun '10

19 Jun '10

Dear all, I am in the process of developing IVR. So just planned to convert my application to handle the TTS voice engine which is supported by FreeSWITCH. Got the mod_unimrcp module which is used to recognize the speech and synthesize the text to voice. For building the mod_unimrcp modules done the following steps. Uncomment the mod_unimrcp line in the modules.conf file in the FreeSWITCH source Given make mod_unimrcp-install command. In modules.conf.xml uncomment the <load modue="mod_unimrcp"/> Configured the following dial plan <extension name="unimrcp"> <condition field="destination_number" expression="^4922$"> <action application="answer"/> <action application="set" data="tts_engine=unimrcp:nuance5-mrcp1"/> <action application="set" data="tts_voice=awb"/> <action application="speak" data="This is our Text to speech on FreeSWITCH"/> --> <action application="sleep" data="500"/> </condition> </extension> While making the call to 4922 got the following error in the FreeSWITCH console. [INFO] mod_dialplan_xml.c:418 Processing thangappan->4922 in context default [NOTICE] mod_dptools.c:717 Channel [sofia/internal/1012(a)192.168.1.222] has been answered [INFO] mod_unimrcp.c:1499 speech_handle: name = unimrcp, rate = 8000, speed = 0, samples = 160, voice = , engine = unimrcp, param = nuance5-mrcp1 [INFO] mod_unimrcp.c:1502 voice = awb, rate = 8000 [NOTICE] mrcp_client.c:549 Create MRCP Handle 0x8b02400 [nuance5-mrcp1] [INFO] mrcp_client_session.c:142 Create Channel 0x8b02400 <new> [INFO] mrcp_client_session.c:398 Receive App Request 0x8b02400 <new> [2] [NOTICE] rtsp_client.c:255 Create RTSP Handle 0x8b04408 [INFO] mrcp_client.c:901 Add MRCP Handle 0x8b02400 [NOTICE] mrcp_client_session.c:718 Add Control Channel 0x8b02400 <new@speechsynth> [INFO] mrcp_client_session.c:420 Send Offer 0x8b02400 <new> [c:0 a:1 v:0] *[ERR] mod_unimrcp.c:965 (TTS-0) Timed out waiting for channel to be ready [ERR] switch_ivr_play_say.c:2104 Invalid TTS module!* [NOTICE] switch_core_state_machine.c:185 sofia/internal/1012(a)192.168.1.222 has executed the last dialplan instruction, hanging up. [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1012(a)192.168.1.222 [CS_EXECUTE] [NORMAL_CLEARING] [NOTICE] switch_core_session.c:1179 Session 1 (sofia/internal/1012(a)192.168.1.222) Ended [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1012(a)192.168.1.222 [CS_DESTROY] [NOTICE] switch_channel.c:669 New Channel sofia/internal/1012(a)192.168.1.222 [49ed5afa-79f0-11df-b531-3553f3a65c3c] So need to find a solution for that. -- Regards, Thangappan.M

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Re: [Freeswitch-users] question about NibbleBill
by Madovsky 18 Jun '10

18 Jun '10

sorry forget my question. I changed the order of the extensions in the default dialplan and set <action application="nibblebill" data="pause"/> and seems to work Thx F ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, June 18, 2010 7:07 PM Subject: question about NibbleBill hi there, I'm trying to create user who can call for free and other not. I set nobal_action in nibblebill conf to an extension in my default dialplan but if the user is at $0 for special case I would authorize the call for free. is it possible ? Thanks F

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Re: [Freeswitch-users] Aastra SCA Woes, Any Updates?
by Brian West 18 Jun '10

18 Jun '10

No it doesn't fully fix it.. again the phone DOES IT WRONG. We hacked FreeSWITCH to guess about it. Its still going to mess up. /b On Jun 17, 2010, at 4:04 PM, Mark Maly wrote: > All, > > Not smart enough to check to the “bit level” but received this from Aastra Tech support earlier and posted to another thread here, but … > > “Your ticket number is xxxxx. > > We have just released a new GA firmware that I would like for you to load onto one of your phones to see if it resolves the issue. You can download this from our website at www.aastratelecom.com/support and click on Download Area and select the 6731i. The firmware version you are looking for is 2.6 and it is listed under Current Software Release. > > Please load that on a test phone and let us know the results. > > Thank you, > > Jessie Fetter > > Aastra Customer Technical Support > support(a)aastra.com > www.aastratelecom.com/support > 800-574-1611 > “ > > It appears this fixed the problems I was encountering, but I’m not comfortable saying it’ll always work. I don’t really have a valid test plan, but SCA “appears” to work with the new firmware…. > > Mark

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Does FS need the pth (gnu pthreads) package?
by Robert Hadley 18 Jun '10

18 Jun '10

Does Freeswitch or the default enabled modules use or need the pthreads library, which on CentOS 5.3 is the pth package? Thanks, Robert

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Merge two calls
by benxmy 18 Jun '10

18 Jun '10

Hi, I'm quite new to freeswitch and voip, so this may be in some way a noob question but I've dug through a lot of the freeswitch docs and done quite a bit of searching and haven't figured it out yet. We're creating a relatively straightforward VoIP system to be used entirely internally (ex: users can only connect with other registered users within our system) and we'd like to be able to combine combine two calls into a single audio stream to the user without the two calls hearing each other. For example, if I'm talking to Mike on line 1 and I'm talking to Erin on line 2, is there a way for me to hear both Erin and Mike simultaneously but for them not to hear each other? Alternatively, is there a straightforward way to simply merge the calls into a conference-type experience where we all hear each other without the user explicitly setting up a conference call? Any and all input is greatly appreciated, as I'm up to my ears in freeswitch but have approximately zero experience with it! Ben -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Merge-two-calls-tp5188763p518… Sent from the freeswitch-users mailing list archive at Nabble.com.

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presence in linksys spa932
by Vladimir Elizarov 18 Jun '10

18 Jun '10

I found a bug in the presence of the linksys spa932 (configure this article: http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932) If during the conversation (lamp on) to the subscriber, someone called, the lamp on the panel goes into the state off. Who can verify whether he has reproduced the same problem? Unit key: fnc=blf+sd+cp;sub=110@$PROXY freeswitch 1.0.6 (git 10 06 2010) Linksys spa962: Software Version: 6.1.3(a) Hardware Version: 1.0.3(917f) Linksys spa932 Unit Enable: Yes Unit Online: Yes Subscribe Expires: 600 Subscribe Retry Interval: 6 HW Version: 1.0.6 SW Version: 2.0.2 Configure image: http://img192.imageshack.us/img192/2320/linksysspa932.png -- Best regards, Vladimir Elizarov

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Re: [Freeswitch-users] How to play a prompt to an ongoing conference?
by David Swardstrom 18 Jun '10

18 Jun '10

The following command is the right command: conference <confname> play <path to file> In JavaScript, the following version of this commanhd should work: apiExecute("conference", confname+" play "+Filename); One word of warning. If you use play to a session, it normally waits until it is done. As I understand it, the above command does not wait. I had to know when it was done, but since I also had to play it to a channel, I decided that they would complete at the same time. There is a separate "conference <> stop" command. I haven't seen anything to suggest a way to determine if something is still playing or recording. I know recording takes up slots in the conference and I suspect that play would also. However, they aren't reported on in a list. Perhaps there should be another version of list, one that reports on the auxiliary channels. Regards, Paul David Swardstrom

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