- FreeSWITCH-dev - FreeSWITCH Mailing Lists

compile error (on windows with VS 2005)
by Dave 09 Sep '08

09 Sep '08

I got the compile error on windows XP pro (with VS 2005). ..\..\src\switch_xml.c(2234) : error C2220: warning treated as error - no 'object' file generated ..\..\src\switch_xml.c(2234) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data ..\..\src\switch_xml.c(2266) : warning C4267: '=' : conversion from 'size_t' to 'unsigned int', possible loss of data ..\..\src\switch_xml.c(2507) : warning C4267: '=' : conversion from 'size_t' to 'unsigned int', possible loss of data ..\..\src\switch_xml.c(2522) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data ..\..\src\switch_xml.c(2529) : warning C4267: '=' : conversion from 'size_t' to 'unsigned int', possible loss of data ..\..\src\switch_xml.c(2600) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data Generating Code... Results Build log was saved at "file://g:\opensource\tele\freeswitch\freeswitch-1.0.1\w32\Library\Debug\BuildLog FreeSwitchCoreLib.htm" FreeSwitchCoreLib - 1 error(s), 6 warning(s) Then there will be lots of compile error since this core library was failed to be built. Please help me out, thanks. ___________________________________________________________ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com.

3 3

Problems with intercept application and attended transfer
by Francisco de Ezcurra 08 Sep '08

08 Sep '08

Hi I'm trying to make an attended transfer from the A leg of a bridge that was created using the intercept application. This works if the transfer is made from the B leg. I've run the tests on a Polycom 501. 1) Dialplan extensions: <extension name="bleg"> <condition field="destination_number" expression="1005"> <action application="set_global" data="leg_b=${uuid}" /> <action application="set" data="hold_music=test/clientdata/system/audios/male/en_US/misc/HoldMusic.wav"/> <action application="gentones" data="${us-ring}|-1" /> </condition> </extension> <extension name="aleg"> <condition field="destination_number" expression="1006"> <action application="set" data="hold_music=test/clientdata/system/audios/male/en_US/misc/HoldMusic.wav"/> <action application="intercept" data="${leg_b}"/> </condition> </extension> <extension name="2003"> <condition field="destination_number" expression="2003"> <action application="set" data="hold_music=test/clientdata/system/audios/male/en_US/misc/HoldMusic.wav"/> <action application="bridge" data="sofia/default/user3.fezcurra"/> </condition> </extension> 2) Steps in which the attended transfer DOES NOT work. 1) With a linksys phone dial the extension 1005 2) With the Polycom 501 dial the extension 1006 which intercept the linksys channel. 3) With the Polycom 501 start an attended transfer to the extension 2003 4) The linksys starts playing the hold music. 5) Answer the extension 2003 6) Confirm the transfer and hang up the Polycom 501. Instead of completed the transfer, FS plays the ring again on the linksys and the extension 2003 stays connected without any bridge. 3) Steps in which the attended transfer works. 1) With the Polycom 501 phone dial the extension 1005 2) With the Linksys dial the extension 1006 which intercepts the Polycom channel. 3) With the Polycom 501 start an attended transfer to the extension 2003 4) The linksys starts playing the hold music. 5) Answer the extension 2003 6) Confirm the transfer and hang up the Polycom 501. 7) The linksys and the extension 2003 are connected properly. Has anybody else experienced this problem ? Thanks Panchi

6 9

usage of <anti-action application="respond" data="407"/>
by Thomas Mangin 07 Sep '08

07 Sep '08

Hello, Could someone tell me if I understand this correctly please. file src/mod/endpoints/mod_sofia/mod_sofia.c line 1021 in case SWITCH_MESSAGE_INDICATE_RESPOND which is what is used by the "respond" function, The domain used to create the challenge is "sip_to_host", which is good when I want to challenge a message DESTINED to the machine itself, however not great if you want to challenge a message to another domain routed through the dialplan like explained on http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_respond . When I am calling 888(a)conference.freeswitch.org, with my phone me(a)domain.com , I challenge the call in the dialplan with a "respond" "407" which sent me back user="me" realm="conference.freeswitch.org" due to sofia_reg_auth_challenge(NULL, tech_pvt->profile, tech_pvt->nh, REG_INVITE, to_host, 0); Is it a bug ? Should it really be "from_host". Can it be changed without breaking existing installation ? Thomas Mangin -- Exa Networks Limited - http://www.exa-networks.co.uk/ Company No. 04922037 - VAT no. 829 1565 09 27-29 Mill Field Road, BD16 1PY, UK Phone: +44 (0) 845 145 1234 - Fax: +44 (0) 1274 567646

1 0

DTMF event being broadcast to different session?
by Johny Kadarisman 05 Sep '08

05 Sep '08

Hi, I'm puzzling with a dtmf problem on my applications. basically, i have a dialplan that run "start_dtmf" and then execute "socket" to my controller apps. * <action application="sleep" data="2000"/> <action application="start_dtmf" data="true"/> <action application="socket" data="192.168.150.52:8084 async full"/> >From the controller applications, I will detect the dtmf event. When the call to system one by one, the application detect dtmf event correctly on each session. But the problem arise when there is 2 or more simultanous call in the system. One caller press digits, and both or all socket sessions receive the dtmf events. I believe this is the cases, but not sure what is the easier way to proof it. Any suggestion on how to troubleshoot or simple way to make sure each sessions don't receive dtmf events. Thanks, Johny K. *

2 6

SIP 180 RINGING w/media in proxy or bypass mode
by Pawel Pierscionek 03 Sep '08

03 Sep '08

Hi, I can easily get ringback (SIP 180 + SDP) when I bridge two SIP calls but can I have one when channel is supposed to proxy media after connecting both legs ? Also is there a way to force ring_ready to send ringback ? Pawel,

2 1

How to configure FreeSWITCH for callingSIPprofile "internal"?
by Евгений Золотов 03 Sep '08

03 Sep '08

We had a question: > How I should reconstruct configuration files of FreeSWITCH, that any user could carry out call at > SIP profile "internal" port 5060 and also it was not required to its registration (message REGISTER from one)? > Like at profiles "external" and "nat" ( version 1.0.trunk( 9377 ) - earlier these profiles were called in another way: > "default" and "outbound"?). ) Brian West has advised to us: >> auth-calls set to false But we know arrangement <param name="auth-calls" value="false"/> in internal.xml, and it works in previous releases. In the previous releases: freeswitch(a)opos8.altron.lan> version FreeSwitch Version 1.0.rc4 (exported) arrangement <param name="auth-calls" value="false"/> in SIP profile default.xml led to successful performance of the command: # ./sipp -sn uac -s 2000 192.168.2.107 (extension 2000 - this is our own extension called application info(), that we also see on console FreeSWITCH). In the last release: freeswitch(a)opos9.altron.lan> version FreeSwitch Version 1.0.trunk (9377) - now the profile is defined in internal.xml, and default - alias - is does not adduce to same result, and INVITE comes to the end with an error UDP message received [644] bytes : SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-4457-1-0;received=192.168.2.107 From: sipp <sip:sipp@127.0.0.1:5061>;tag=4457SIPpTag001 To: sut <sip:2000@192.168.2.107:5060>;tag=ayUKNQ761p51r Call-ID: 1-4457(a)127.0.0.1 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9377 And we can see such message on the FreeSWITCH console? 2008-09-03 11:35:57 [WARNING] sofia.c:3158 sofia_handle_sip_i_invite() IP 192.168.2.107 Rejected by acl domains Why does it happens? ( May be this is mistakes in a new version?) Even if we have been change standart scenario SIPP to a call form a user "1000" $ ./sipp -sd uac > uac.xml ... changing name "sipp" to a "1000" everywhere ... $ ./sipp -sf uac-1000.xml -s 2000 -m 1 192.168.2.107 -trace_msg - we have the same result. How do we can get the same result, that inti version 1.0 rc4?

1 0

INFO: wait_for_silence wikified; set chan var w/ result?
by David Brazier 02 Sep '08

02 Sep '08

Hi A return value would be very useful - did you get any further with this? Also, I'm puzzled by the "countdown" in switch_ivr_wait_for_silence - it just adds an arbitrary 25 frame (0.5sec) delay to completing the call. Anny help appreciated. David MC Wrote: > FYI, > > I am working on the return values for this function. I've settled on a > variable name of ${waitstatus} - unless there are any objections. > > The return values I've got so far are: > "silence" > "timeout" > "error" > > I am looking at making the error return value more meaningful but for > now since this is so new I think the above three items will get us > going. I'll submit the patch as soon as it is ready. 360CRM Limited, registered address, Caledonia House, 89 Seaward Street, Glasgow, G41 1HJ, Registered in Scotland, company number SC261952. This email and its attachments may be confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of 360CRM Ltd. This email does not form any part of a contract. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error.

3 2

Re: [Freeswitch-dev] How to save RTP-audiostrem to a file?
by ??????? ??????? 02 Sep '08

02 Sep '08

On Monday, 1 September 2008 12:14 Brian West <brian(a)freeswitch.org> wrote: > FreeSWITCH will always be in the media path unless you set bypass_media. > > /b > > On Sep 1, 2008, at 12:06 PM, ??????? ??????? wrote: > >> What values need to be set in variables that all RTP traffic passed >> through FreeSWITCH? >> > >

3 2

Flite additional voices
by Frank - IMPACT 01 Sep '08

01 Sep '08

I would like to use some of the other voices from festival other than Kal with the Diphone method. This seems to be the only voice that is automatically build for FS. Has anyone come up with the procedures to build and add in new voices from Festival for use in FS?

2 2

How to save RTP-audiostrem to a file?
by Евгений Золотов 01 Sep '08

01 Sep '08

We had a question: > With which application and API is possible to keep full content of RTP audiostream to a file? > Our FreeSWITCH is configured in such manner that RTP-traffic directing throw FreeSWITCH, > but not between UAs. And took answer on it: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session ------------------------------------------------------------------- On it HELP we have mase such extension : <extension name="ext-record"> <condition field="destination_number" expression="^4010$"> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2008, Altron Inc." /> <action application="set" data="RECORD_SOFTWARE=FreeSwitch" /> <action application="set" data="RECORD_ARTIST=Registration" /> <action application="set" data="RECORD_COMMENT=!!!" /> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="sofia/nat/${dialed_ext}@localhost:5062" /> <action application="answer" /> </condition> </extension> This draft works already, but there are new questions: 1. It's seems, that record of RTP session can occur only when RTP stream is routed through FS: UA-> FS-> UA, instead of directly (in LAN) UA-> UA. For this purpose, as we understand, it is necessary to set values in variables of FreeSWITCH: <action application="set" data="bypass_media=..."/> <action application="set" data="proxy_media=..."/> <action application="set" data="inbound_late_negotiation=..."/> ( for SIP profile nat for extension ext-record). What values need to be set in variables that all RTP traffic passed through FreeSWITCH? 2. At extension ext-record we set record_session then we do bridge to leg B. As it is possible to configurate registration from another extension, for example: <extension name="XXXX"> <condition field="destination_number" expression="^XXXX$"> ... <action application="execute_extension" data="ext-record XML public "/> ... <action application="bridge" data="sofia/nat/YYYY" /> ... </condition> </extension> - where own extension not call any leg B, but only record session XXXX->YYYY. How we are able to do this??? Thankful in advance, Evgeniy.

2 1