- FreeSWITCH-users - FreeSWITCH Mailing Lists

VegaStream Contacts
by Michael Collins 18 Aug '11

18 Aug '11

Hello all, If you know someone who works at VegaStream please contact me off list. I need to contact VegaStream and I prefer not to go in cold if I can help it. Thanks, Michael

1 0

g.729 licence usage
by Julien Chavanton 18 Aug '11

18 Aug '11

It seems 2 calls bridged with recording consume 3 licences ? (2 encoders and 3 decoders) Is there a way to minimize the usage, I do not want DTMF capture (inband) but only need to record. freeswitch@internal> show calls call_uuid,call_created,call_created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid,hostname e0567680-8c11-4cce-8462-b277f8632d0e,2011-08-18 18:31:12,1313692272,switch_ivr_multi_threaded_bridge,Outbound Call,1514xxxxxxx,xxxxxxxxx,sofia/external/1514xxxxxxx,e0567680-8c11-4cce-8462-b277f8632d0e,Outbound Call,2xxxxxxxxxx,b_leg_answer,sofia/external/2xxxxxxxxxx,419da0d6-7723-4f2c-b856-20c73cbd91c1,voxcast01 1 total. freeswitch@internal> g729_used 2:3

1 0

RFC2833 and Inband DTMF
by Sam 18 Aug '11

18 Aug '11

Hi, I wanted to know if Freeswitch offered a dtmf mode to negotiate whether the inbound/A-leg's dtmf is rfc2833 or inband compatible? Just like how Asterisk has "dtmfmode=auto".

1 0

Exchanging IMs via RTMP
by afshin afzali 18 Aug '11

18 Aug '11

Hi FreeSWITCHers, I think about a chat service which an anonymous be able to send and receive IM to / from a selected user. Based on the flex client and the server side code it seems that sending SEND_MESSAGE event can handle half of this scenario. My questions is : if the anonymous, registers itself as a fake / temporary user, will be able to receive IM as well ? It seems that onEvent function does not invoke in netStatus fuction ! BEST, -- afshin

1 0

Great ClueCon!
by Peter Olsson 18 Aug '11

18 Aug '11

Hi all, I just wanted to thank everyone involved in the ClueCon conference. It was the first time for me, but definately not the last one :) It was very well organized, and it was really great to meet some of the people I usually just "talk" to on the mailing list and Jira etc. /Peter

2 1

Simulating Asterisk dial(x:y:z)
by Sam 18 Aug '11

18 Aug '11

I want to simulate the Asterisk dial(x:y:z) app where (x=total call time, y=warning time, z=warning interval) in Freeswitch. I know it can be done with sched_hangup and sched_broadcast, my question is how can I call a series of "api_on_answer" comands to do it?

3 4

Problem with FIFO music and chime
by Tomasz Hyziak 18 Aug '11

18 Aug '11

Hi I created simple IVR. When caller presses 2, he is redirected to FIFO. But i've got a problem with fifo_music and fifo_chime_* variables... When I set only fifo_music - music plays. When I set fifo_music and fifo_chime_freq (set to 10) and fifo_chime_list - music play ONLY. There are no chime every 10 seconds. When I set only fifo_chime_* variables (without fifo_music) - chime plays every 10 seconds... I've got no idea why it happend - it was working about 2 weeks ago... I use FreeSwitch from git (downloaded @ 4 July). Dialplan: <extension name="in_dialog_1"> <condition field="destination_number" expression="^(hidden)$"> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="/srv/nagrania/aktualne/IN_${strftime(%Y%m%d-%H%M%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="set" data="media_bug_answer_req=true"/> <action application="set" data="domain_name=$${domain}"/> <action application="export" data="absolute_codec_string=PCMU,PCMA"/> <action application="export" data="codec_string=PCMU,PCMA"/> <action application="set" data="jitterbuffer_msec=180"/> <action application="set" data="hangup_after_bridge=true"/> <action application="answer"/> <action application="sleep" data="2000"/> <action application="set" data="fifo_chime_list=/srv/nagrania/ivr/prosze_czekac.wav"/> <action application="set" data="fifo_chime_freq=10"/> <action application="set" data="fifo_music=/usr/local/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav"/> <action application="ivr" data="ivr"/> <action application="fifo" data="fifo_ogolne in"/> <action application="hangup"/> </condition> </extension> IVR: <include> <menu name="ivr" greet-long="/srv/nagrania/ivr/powitanie_pelne.wav" greet-short="/srv/nagrania/ivr/powitanie_skrocone.wav" invalid-sound="/srv/nagrania/ivr/zla_opcja.wav" exit-sound="/srv/nagrania/ivr/laczenie_z_operatorem.wav" confirm-macro="" confirm-key="" tts-engine="flite" tts-voice="rms" confirm-attempts="2" timeout="5000" inter-digit-timeout="2000" max-failures="2" max-timeouts="2" digit-len="1"> <entry action="menu-exec-app" digits="1" param="bridge user/1100"/> <entry action="menu-exec-app" digits="2" param="fifo fifo_ogolne in"/> <entry action="menu-top" digits="9"/> </menu> </include> FIFO: <configuration name="fifo.conf" description="FIFO Configuration"> <settings> <param name="delete-all-outbound-member-on-startup" value="false"/> </settings> <fifos> <fifo name="fifo_ogolne" importance="0"> <member timeout="10" simo="1" lag="2">{fifo_member_wait=nowait}user/1110</member> <member timeout="10" simo="1" lag="2">{fifo_member_wait=nowait}user/1111</member> <member timeout="10" simo="1" lag="2">{fifo_member_wait=nowait}user/1112</member> ...other users... </fifo> </fifos> </configuration> -- Greetings - Tomasz Hyziak

2 3

Audio Problem - one way
by Erin O'Meara 18 Aug '11

18 Aug '11

I have to VOIP servers in the cloud(same provider), one is FusionPBX running Freeswitch the other is Elastix with Asterisk. I have two phones(same Grandstream GXP280) in my office each registered to one of the servers. I have one phone number and when I route it thru the Elastix server, I get no complaints about the audio. When I route the number thru the FusionPBX the person on the other end complains of choppy audio but I can't tell the difference on my end. But when I call from my cell I can tell that the Freeswitch server audio is not as good on the outgoing. Both Virtual servers are the same specs, memory and proc. The elastix box has more overhead with all the extra crap it runs and I can look at the performance metric's on the virtual servers and confirm that the fusionpbx has less proc usage and memory usage. The calls are using the same codec (ulaw). Currently the call volume is just my one call for the Fusion server, and at most 2 for the Elastix. Any thoughts on with the same setup and internet routes, audio in one direction is preforming so poorly with the freeswitch? Regards, 206.905.9520 http://salmonbaytechnology.com

1 0

voicemail is not saved
by Yungwei Chen 18 Aug '11

18 Aug '11

Hi, I left several voicemails (Each is longer than 3 sec) to a user account, but none is available when I check the mailbox. Relevant settings are listed below. What am I missing here? Thanks. In conf/autoload_configs/modules.conf.xml, mod_voicemail is already loaded. freeswitch@internal> load mod_voicemail +OK Reloading XML -ERR [Module already loaded] freeswitch@internal> 2011-08-15 14:32:10.666978 [WARNING] switch_loadable_module.c:998 Module mod_voicemail Already Loaded! Here's the content of conf/autoload_configs/voicemail.conf.xml: <configuration name="voicemail.conf" description="Voicemail"> <settings> </settings> <profiles> <profile name="default"> <param name="file-extension" value="wav"/> <param name="terminator-key" value="#"/> <param name="max-login-attempts" value="3"/> <param name="digit-timeout" value="10000"/> <param name="min-record-len" value="3"/> <param name="max-record-len" value="300"/> <param name="max-retries" value="3"/> <param name="tone-spec" value="%(1000, 0, 640)"/> <param name="callback-dialplan" value="XML"/> <param name="callback-context" value="default"/> <param name="play-new-messages-key" value="1"/> <param name="play-saved-messages-key" value="2"/> <param name="login-keys" value="0"/> <param name="main-menu-key" value="0"/> <param name="config-menu-key" value="5"/> <param name="record-greeting-key" value="1"/> <param name="choose-greeting-key" value="2"/> <param name="change-pass-key" value="6"/> <param name="record-name-key" value="3"/> <param name="record-file-key" value="3"/> <param name="listen-file-key" value="1"/> <param name="save-file-key" value="2"/> <param name="delete-file-key" value="7"/> <param name="undelete-file-key" value="8"/> <param name="email-key" value="4"/> <param name="pause-key" value="0"/> <param name="restart-key" value="1"/> <param name="ff-key" value="6"/> <param name="rew-key" value="4"/> <param name="skip-greet-key" value="#"/> <param name="previous-message-key" value="1"/> <param name="next-message-key" value="3"/> <param name="skip-info-key" value="*"/> <param name="repeat-message-key" value="0"/> <param name="record-silence-threshold" value="200"/> <param name="record-silence-hits" value="2"/> <param name="web-template-file" value="web-vm.tpl"/>    <param name="operator-extension" value="operator XML default"/> <param name="operator-key" value="9"/> <param name="vmain-extension" value="vmain XML default"/> <param name="vmain-key" value="*"/>   <email> <param name="template-file" value="voicemail.tpl"/> <param name="notify-template-file" value="notify-voicemail.tpl"/>  <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/> <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/> </email>      </profile> </profiles> </configuration> In conf/directory/default.xml, user 91000 is defined in domain voicemail_2. <domain name="voicemail_2"> <groups> <group name="default"> <users> <user bluebox_id="2" id="91000"> <params> <param name="sip-forbid-register" value="true"/> <param name="vm-password" value="xxxx"/> <param name="vm-message-ext" value="wav"/> <param name="vm-mailto" value="xxx(a)xxx.com"/> <param name="email-addr" value="xxx(a)xxx.com"/> <param name="vm-delete-file" value="false"/> <param name="vm-attach-file" value="true"/> <param name="vm-email-all-messages" value="true"/> </params> <variables> <variable name="timezone" value="America/Chicago"/> </variables> </user> </users> </group> </groups> </domain> In my dialplan, calls to 1112223333 will be sent to user 91000's voicemail box if they are not answered. <extension name="main_number_2" continue="true"> <condition field="destination_number" expression="^1112223333$"> <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/> <action application="set" bluebox="settingFail" data="continue_on_fail=true"/> <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/> <action application="set" bluebox="settingTimeout" data="call_timeout=30"/> <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/> <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/> <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=m1"/> <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=1112223333"/> <action application="bridge" data="user/91000@$${location_3}"/> <action application="answer"/> <action application="voicemail" data="default voicemail_2 91000"/> <action application="hangup"/> </condition> </extension>

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Original Caller ID/number after attended transfer
by Weigel, Stefan 18 Aug '11

18 Aug '11

0 *H÷ 010 +0 *H÷ $ Content-Type: multipart/alternative; boundary="----=_NextPart_000_0C65_01CC5DB3.B4FD4C90" Dies ist eine mehrteilige Nachricht im MIME-Format. ------=_NextPart_000_0C65_01CC5DB3.B4FD4C90 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Hi all, is there a possibility to display the original caller ID & number after doing a attended transfer. External call to phone A -> calls phone B (I see caller ID & number of phone A) -> doing an attended transfer of external call to phone B (still caller ID & number of phone A). Thanks in advance and best regards, Stefan ------=_NextPart_000_0C65_01CC5DB3.B4FD4C90 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable m<html xmlns:v=3D"urn:schemas-microsoft-com:vml" = xmlns:o=3D"urn:schemas-microsoft-com:office:office" = xmlns:w=3D"urn:schemas-microsoft-com:office:word" = xmlns:x=3D"urn:schemas-microsoft-com:office:excel" = xmlns:p=3D"urn:schemas-microsoft-com:office:powerpoint" = xmlns:a=3D"urn:schemas-microsoft-com:office:access" = xmlns:dt=3D"uuid:C2F41010-65B3-11d1-A29F-00AA00C14882" = xmlns:s=3D"uuid:BDC6E3F0-6DA3-11d1-A2A3-00AA00C14882" = xmlns:rs=3D"urn:schemas-microsoft-com:rowset" xmlns:z=3D"#RowsetSchema" = xmlns:b=3D"urn:schemas-microsoft-com:office:publisher" = xmlns:ss=3D"urn:schemas-microsoft-com:office:spreadsheet" = xmlns:c=3D"urn:schemas-microsoft-com:office:component:spreadsheet" = xmlns:odc=3D"urn:schemas-microsoft-com:office:odc" = xmlns:oa=3D"urn:schemas-microsoft-com:office:activation" = xmlns:html=3D"http://www.w3.org/TR/REC-html40" = xmlns:q=3D"http://schemas.xmlsoap.org/soap/envelope/" = xmlns:rtc=3D"http://microsoft.com/officenet/conferencing" = xmlns:D=3D"DAV:" xmlns:Repl=3D"http://schemas.microsoft.com/repl/" = xmlns:mt=3D"http://schemas.microsoft.com/sharepoint/soap/meetings/" = xmlns:x2=3D"http://schemas.microsoft.com/office/excel/2003/xml" = xmlns:ppda=3D"http://www.passport.com/NameSpace.xsd" = xmlns:ois=3D"http://schemas.microsoft.com/sharepoint/soap/ois/" = xmlns:dir=3D"http://schemas.microsoft.com/sharepoint/soap/directory/" = xmlns:ds=3D"http://www.w3.org/2000/09/xmldsig#" = xmlns:dsp=3D"http://schemas.microsoft.com/sharepoint/dsp" = xmlns:udc=3D"http://schemas.microsoft.com/data/udc" = xmlns:xsd=3D"http://www.w3.org/2001/XMLSchema" = xmlns:sub=3D"http://schemas.microsoft.com/sharepoint/soap/2002/1/alerts/"= xmlns:ec=3D"http://www.w3.org/2001/04/xmlenc#" = xmlns:sp=3D"http://schemas.microsoft.com/sharepoint/" = xmlns:sps=3D"http://schemas.microsoft.com/sharepoint/soap/" = xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance" = xmlns:udcs=3D"http://schemas.microsoft.com/data/udc/soap" = xmlns:udcxf=3D"http://schemas.microsoft.com/data/udc/xmlfile" = xmlns:udcp2p=3D"http://schemas.microsoft.com/data/udc/parttopart" = xmlns:wf=3D"http://schemas.microsoft.com/sharepoint/soap/workflow/" = xmlns:dsss=3D"http://schemas.microsoft.com/office/2006/digsig-setup" = xmlns:dssi=3D"http://schemas.microsoft.com/office/2006/digsig" = xmlns:mdssi=3D"http://schemas.openxmlformats.org/package/2006/digital-sig= nature" = xmlns:mver=3D"http://schemas.openxmlformats.org/markup-compatibility/2006= " xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" = xmlns:mrels=3D"http://schemas.openxmlformats.org/package/2006/relationshi= ps" xmlns:spwp=3D"http://microsoft.com/sharepoint/webpartpages" = xmlns:ex12t=3D"http://schemas.microsoft.com/exchange/services/2006/types"= = xmlns:ex12m=3D"http://schemas.microsoft.com/exchange/services/2006/messag= es" = xmlns:pptsl=3D"http://schemas.microsoft.com/sharepoint/soap/SlideLibrary/= " = xmlns:spsl=3D"http://microsoft.com/webservices/SharePointPortalServer/Pub= lishedLinksService" xmlns:Z=3D"urn:schemas-microsoft-com:" = xmlns:st=3D"" xmlns=3D"http://www.w3.org/TR/REC-html40"> <head> <META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; = charset=3Dus-ascii"> <meta name=3DGenerator content=3D"Microsoft Word 12 (filtered medium)"> <style>  </style>  </head> <body lang=3DDE link=3Dblue vlink=3Dpurple> <div class=3DSection1> Hi all,<o:p></o:p> <o:p> </o:p> is there a possibility to = display the original caller ID & number after doing a attended transfer. = External call to phone A -> calls phone B (I see caller ID & number of phone A) = -> doing an attended transfer of external call to phone B (still caller ID & = number of phone A).<o:p></o:p> <o:p> </o:p> <o:p> </o:p> Thanks in advance and best = regards,<o:p></o:p> <o:p> </o:p> Stefan<o:p></o:p> </div> </body> </html> 1 ------=_NextPart_000_0C65_01CC5DB3.B4FD4C90-- 20é0Ñ £0 *H÷ 0>10 UDE10U Allianz Group10UAllianz UserCA0 100825090218Z 120826090218Z0j10 UDE10U Allianz110/ *H÷ "stefan.weigel(a)allianz-warranty.com10U Stefan Weigel0"0 *H÷ 0 ^umÉ/MOõ88)AfN8÷ÒpD,Æí ÃI-Ra±«÷êÕJ8ðäðÞ\RI.Doo4Ê8Ð7#{ÃL8.}D¢0¬µÊ®íà~¥L172ÔùPUê©Áv×Ð2"2;ºd€xt/®0Yï8¶Fd^Æ<wÝCbÌgÊ[üOd^€B2#âõß¬{.ÁØÅT|¯;éY1 Î#¥hnñéÜ*uz©vÛIöG&aú>±ÅÛèŒ¿S¶:]âê³€qÜòí-¹_=¢^LÁûë-*Y£Ã0À0Uÿ00U#0ÞÄÖÿzHÀs¿þo²ôõøl0U%0 +0-U&0$"stefan.weigel(a)allianz-warranty.com0IU B0@0> +·w000.+"https://rootca.allianz.com/userca/0 *H÷ Æ±G-?Áé 0[kÖ1Ìõ0Æ9 ÑrG¯&sk VïpcìþÀ:íÅùÎÀ`ôGÜ|Êñœ;Ø3ºÞÈêIòó*ÁœjdQC8ËÎsª_Ââ®u&Yr¢»ÎV4' Oê©êµ¬¡×X÷ßn>±äÍ @Oþ?<zÿ %oÐTÌÂW~ 72I8[³ÕVMöTbÙc±éÍ³h³ŠØRNq*Ü÷÷úêºînçßºú`³]^èYEl,Ûš°¶oXr>¶ÑRH$bËþ:%tôµÑŽs@ëÝû0,0 ±l0 *H÷ 0>10 UDE10U Allianz Group10UAllianz UserCA0 110302152445Z 130303152445Z0j10 UDE10U Allianz110/ *H÷ "stefan.weigel(a)allianz-warranty.com10U Stefan Weigel0"0 *H÷ 0 ŠÄãZù«V€Xð,Êd©dP²b¹Ux^ôiDª#î³z0éarm e²2QšcC,¡ëó€}ÿåÊhWÌ÷~cPØ&ÚlÕõÒSv"µUrŠÏx^HÞèy[M±fòFÛ^j,·ñUÂ³_co;5O¬{§PÑ³Þ ØèJŒa.ŸÅÎÕ®ÕKÔ9ñËäÞÄÈoETbÔuütNg«,è7ÖæTý&µeâöïüªÀú²RxM§4tùîS¥ÉC»Ô$¹"3áLfÞEfNÒ¿$¡)øß£00Uÿ0U#0ÞÄÖÿzHÀs¿þo²ôõøl0U%0++05U.0,0* ( &$http://rootca.allianz.com/userca.crl0IU B0@0> +·w000.+"https://rootca.allianz.com/userca/0-U&0$"stefan.weigel(a)allianz-warranty.com0 *H÷ [ &þ\pQ$:HÙ,t^ßyùònó_A jCÔÀÜqV,oú; 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