April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Build fails on Suse
by Matt White 29 Apr '10

29 Apr '10

I'm using the opensuse build service to create rpm's for Suse. The build fails with the following error below. I think its due to the gcc 4.3 used on SLE11 as I can't replicate it in older versions I: Expression compares a char* pointer with a string literal. Usually a strcmp() was intended by the programmer E: freeswitch stringcompare strings/apr_snprintf.c:1261 Any thoughts? I'm using the nightly snapshot. -M

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effective_caller_id_name not working while calling multiple endpoints with :_:
by Peter P GMX 29 Apr '10

29 Apr '10

Setting the effective_caller_id_name when dialing multiple endpoints with :_: do not seem to work. See example: <action application="set" data="effective_caller_id_name=MyName"/> <action application="bridge" data="user/30@mydomain.com:_:user/31@mydomain.com:_:user/32@mydomain.com:_:user/33@mydomain.com:_:user/34@mydomain.com"/> Freeswitch tries to set it: EXECUTE sofia/external/069xxxxxxxx@10.xx.xx.1414:5060 set(effective_caller_id_name=MyName) 2010-04-21 11:11:48.642571 [DEBUG] mod_dptools.c:816 sofia/external/069xxxxxxxx@10.xx.xx.1414:5060 SET [effective_caller_id_name]=[MyName] But the SIP messages do not contain the effective_caller_id_name. If we change the ":_:" sperator to "," then the effective_caller_id_name is correctly submittted (hower I cannot call multiple-registrations on one number then). We are on FreeSWITCH Version 1.0.head (svn-17188) Any ideas how to overcome this? Or shall I open a JIRA? Best regards Peter See example SIP message: U 10.xx.xx.141:5060 -> 10.xx.xx.14172:2048 INVITE sip:31@10.xx.xx.14172:2048;line=hxbudrul SIP/2.0. Via: SIP/2.0/UDP 10.xx.xx.141;rport;branch=z9hG4bKZD66c84339SHH. Max-Forwards: 70. From: "069xxxxxxxx" <sip:069xxxxxxxx@10.xx.xx.141>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.14172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@10.xx.xx.141:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 920. X-FS-Support: update_display. Remote-Party-ID: "069xxxxxxxx" <sip:069xxxxxxxx@10.xx.xx.141>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.141. s=FreeSWITCH. c=IN IP4 10.xx.xx.141. t=0 0. m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9 121 3 101 13. a=rtpmap:115 G7221/32000. a=fmtp:115 bitrate=48000. a=rtpmap:96 AMR/8000. a=fmtp:96 octet-align=0. a=rtpmap:99 SPEEX/32000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:7 LPC/8000. a=rtpmap:124 G726-16/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:6 DVI4/16000. a=rtpmap:123 G726-24/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:10 L16/22050. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode #

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mod_shout recordFile bitrate
by Andy 29 Apr '10

29 Apr '10

Hi folks, I'm using session.recordFile to push the audio from my calls to an icecast server. As far as I can tell the bitrate of the audio I'm sending is only 64kbps but the info sent to icecast about the stream always says 24000kbps which is causing me problems. Is there anyway I can correct the value sent in the info packet without editing the source code of mod_shout? This may be a question for the developers list but, If I did decide to edit the code in mod_shout.c is there a way I can set this to the actual bit rate for the stream rather than hardcoding it to 64kbps which is actually always true with my application. Cheers Andy

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mod_shout error on recording session!
by babak yakhchali 29 Apr '10

29 Apr '10

Hi I'm using mod_shout in mod_managed to record calls in mp3 format but after 4 or 5 seconds I get this: 2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 LAME 3.97 32bits ( http://www.m p3dev.org/) *Assertion failed: enn >= 0, file ..\..\lame-3.97\libmp3lame\psymodel.c, line 497* 2010-04-29 11:24:56.500000 [INFO] mod_shout.c:297 Using polyphase lowpass filter , transition band: 3903 Hz - 4000 Hz

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Intercom w/ Mute
by Gabriel Kuri 29 Apr '10

29 Apr '10

Is there support for dialing an extension and automatically answering and muting the dialed extension? I found this page on the wiki, but it doesn't mention anything about automaticaly muting the call? http://wiki.freeswitch.org/wiki/Intercom Cheers, Gabe

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build error
by babak yakhchali 29 Apr '10

29 Apr '10

I'm trying to build the lates svn and I got these errors 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_hash' : undeclared identifier 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_d' : undeclared identifier 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_e' : undeclared identifier 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]' 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]' 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t' 1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065: 'msg_unknown_hash' : undeclared identifier

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Functionality
by Sean Holt 29 Apr '10

29 Apr '10

Hello list, I¹ve been looking through the wiki to figure the best way to enable the receptionist to turn on or off an ivr after hours auto-attendant. I¹m thinking the receptionist can enter a couple digits to control this functionality. Can someone provide an example or show me in the wiki a good place to figure this out Thanks Sean

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say phrase into recordFile using lua script
by Jesse Cloutier 28 Apr '10

28 Apr '10

Hello, I am having trouble with a lua script that i am trying to put together. The script basically makes a recording using the "session:recordFile()" function, and is terminated by a callback function that returns "break" on dtmf "#" What I would like to do is have the system say the date and time ( session:say() ) into the recording at the end of the recording. It should be triggered by the call back. When I try it out it works great, I hear the system say the date / time, except that it is never recorded into the recording. I have been searching the docs for something that might clear up why its not recording but I can't find anything. I thought maybe doing a uuid_broadcast would fix the problem but there is no builtin function for that in the lua api and I was not able to make it work using the "session:execute" function either. Any help would be really appreaciated! This is the relavent code, for testing purposes I am having the date spoken using an alternative digit, you can also see my attempt to do a uuid_broadcast and sched_broadcast commented out: ###################### --call back function function onInputRecord(s, type, obj) -- function to end recording if (type == "dtmf") then if ( obj['digit'] == '1') then curDate = os.date(); session:say(curDate, "en", "CURRENT_DATE_TIME", "pronounced"); --uuidBroadcast = " say::en CURRENT_DATE_TIME pronounced " .. os.date() .. " both"; --session:execute("uuid_broadcast ".. UUID, uuidBroadcast); --session:execute("sched_broadcast", "data=\"+1 /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-hello.wav both\""); return true; elseif ( obj['digit'] == '#' ) then return "break"; end end end session:setInputCallback("onInputRecord", ""); RECORD_PATH = /parth/to/file session:recordFile(RECORD_PATH, 30000, 10, 10); -- pressing # ends the recording #################### Thanks alot! JC

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Having a problem with attended transfer when FS is the transfer target
by Mardy Marshall 28 Apr '10

28 Apr '10

I'm having a problem with attended transfers where the destination of the transfer is a FreeSWITCH based application such as FSComm. (It should be noted that in my setup the phone performing the transfer and the caller which is being transferred are parties of another SIP server.) What I see, from a SIP signaling standpoint, is that after FreeSWITCH receives and acknowledges the INVITE w/Replaces it does not terminate the initial call leg by sending a BYE to the transfer controller. From the FreeSWITCH application side, FS still thinks that both the initial call leg and transferred call leg are active. I experimented with trying to explicitly terminate the initial call leg by using uuid_kill, but this caused FS to kill all legs of the call. Is there a specific action that the application must take in order for the transfer to complete? -Mardy

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FreeSWITCH Conference Call Starting Shortly!
by Michael Collins 28 Apr '10

28 Apr '10

Come join us! http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_28 -MC

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