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One-way audio issue with attended transfer
by John Platts 07 Sep '11

07 Sep '11

We have been experiencing a one-way audio issue with attended transfer. The problem is that after an call is transferred through an attended transfer, the caller cannot be heard, but the caller can hear the voice of the person that the call is transferred to. Steps to reproduce the problem: 1. Dial into FreeSWITCH from an external phone. 2. Answer the call. One-way audio is not experienced here. 3. Press the Transfer button on the IP phone to initiate an attended transfer. 4. Dial the extension that the call is to be transferred to. 5. Allow the extension that the call is to be transferred to pick up the call. One-way audio is not experienced here. 6. Press the Transfer button to complete the transfer. The caller does hear audio from the person that the call is transferred to, but the person that the call is transferred to cannot hear the caller. The one way audio issue is occurring here. The attended transfer was performed from an extension on one SPA303 phone to an extension on another SPA303 phone. FreeSWITCH actually detects that the transfer is an attended transfer. Here are attachments that I have uploaded to pastebin: conf/dialplan/features.xml file: http://pastebin.freeswitch.org/17291 FreeSWITCH log, including the attended transfer: http://pastebin.freeswitch.org/17286 How do we get this one-way audio issue fixed?

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Bug or Not? "longjmp causes uninitialized stack frame"
by Michael Toop 07 Sep '11

07 Sep '11

Hi, Sorry not sure what to do with this, if this is a bug or another problem, but getting this message in the bt when running FS as it does a core dump: " longjmp causes uninitialized stack frame". Running on: 2.6.38-11-server #48-Ubuntu, Ubuntu 11.04 on the latest GIT release. Thanks, Michael 2011-09-04 15:34:59.149089 [INFO] mod_native_file.c:94 Opening File [/usr/local/freeswitch/sounds/custom/mwc-enter-destination.G729] 8000hz 2011-09-04 15:34:59.169011 [INFO] sofia.c:755 sofia/cellc/ 763070366(a)172.103.0.36 Update Callee ID to "763070366" <763070366> 2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1367 Session 946 (sofia/cellc/790938072(a)172.103.0.36) Ended 2011-09-04 15:34:59.169011 [NOTICE] switch_core_session.c:1369 Close Channel sofia/cellc/790938072(a)172.103.0.36 [CS_DESTROY] *** longjmp causes uninitialized stack frame ***: freeswitch terminated ======= Backtrace: ========= /lib/x86_64-linux-gnu/libc.so.6(__fortify_fail+0x37)[0x7f1157fbe1d7] /lib/x86_64-linux-gnu/libc.so.6(+0xfe169)[0x7f1157fbe169] /lib/x86_64-linux-gnu/libc.so.6(__longjmp_chk+0x33)[0x7f1157fbe0d3] /usr/lib/libcurl.so.4(+0xd165)[0x7f1156457165] /lib/x86_64-linux-gnu/libpthread.so.0(+0xfc60)[0x7f1158a89c60] /lib/x86_64-linux-gnu/libc.so.6(__select+0x33)[0x7f1157f9e143] /usr/local/freeswitch/lib/libfreeswitch.so.1(apr_sleep+0x45)[0x7f11593d16c5] /usr/local/freeswitch/lib/libfreeswitch.so.1(+0xcab5c)[0x7f11593a7b5c] /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_console_loop+0x74a)[0x7f1159332e9a] freeswitch[0x402b22] /lib/x86_64-linux-gnu/libc.so.6(__libc_start_main+0xff)[0x7f1157edeeff] freeswitch[0x401769] ======= Memory map: ======== 00400000-00404000 r-xp 00000000 08:01 1845708 /usr/local/freeswitch/bin/freeswitch 00603000-00604000 r--p 00003000 08:01 1845708 /usr/local/freeswitch/bin/freeswitch 00604000-00605000 rw-p 00004000 08:01 1845708 /usr/local/freeswitch/bin/freeswitch 017b0000-026c9000 rw-p 00000000 00:00 0 [heap] 7f114004d000-7f114004e000 ---p 00000000 00:00 0 7f114004e000-7f1140089000 rw-p 00000000 00:00 0 7f1140089000-7f114008a000 ---p 00000000 00:00 0 7f114008a000-7f11400c5000 rw-p 00000000 00:00 0

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mod_fifo to Voicemail if there's no "agents"
by Silvio Escher 07 Sep '11

07 Sep '11

Hi there, ive following in my public dp: <extension name="zentrale_fifo"> <condition field="destination_number" expression="^zentrale_fifo$"> <action application="set" data="fifo_music=intro_und_schleife.wav"/> <action application="answer"/> <action application="set" data="fifo_orbit_exten=zentrale_voicemail:30"/> <action application="fifo" data="zentrale_fifo in"/> </condition> </extension> <extension name="Zentrale"> <condition field="${sip_to_user}" expression="^0631680350$" break="on-false"/> <condition wday="2-6" hour="9-16"> # zeit geht bis 59 <action application="transfer" data="zentrale_fifo XML public"/> <anti-action application="answer"/> <anti-action application="sleep" data="1500"/> <anti-action application="playback" data="niemand_da_zeit.wav"/> <anti-action application="sleep" data="1000"/> <anti-action application="hangup" data="NO_ANSWER"/> </condition> </extension> and iam adding Members to the fifo at runtime ( during *extension or an daily reset by cron ) ./fs_cli -x "fifo_member add zentrale_fifo user/26" or ./fs_cli -x "fifo_member del zentrale_fifo user/26" Everything works fine beside the little issue that if theres no Member/Agent in the fifo - the Call is just also waiting 30 seconds till transfered to voicemail. I noticed that with fifo list zentrale_fifo iam able to get an Memberlist ( fifo count just shows me 0 Members - dunno why - bug ? ) - but iam unsure how to process further. Whats the ideal Solution to get an Caller directly to the Voicemail when no Member/Agent is in the called Fifo ? Best Regards, Silvio -- Silvio Escher omeco GmbH

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Problem receiving fax
by Antonio 07 Sep '11

07 Sep '11

Hi to all, I'm having problems receiving fax in a pri E1 line. When a restart freeswitch or the server and i can receive one fax and then it starts to fail again. I didn't open a jira bug, because I'm not sure where to look for the problem. The log can be found at http://pastebin.freeswitch.org/17047 Thanks, António Silva

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DTMF from Nortel BCM 400 to Freeswitch
by Ray Pang 07 Sep '11

07 Sep '11

I’ve been unable to get DTMF to work from BCM 400 to Freeswitch PSTN -> BCM -> sip forward -> Freeswitch Autoattendant. DTMF never gets recognized: 2011-08-30 13:13:46.164796 [DEBUG] switch_ivr_menu.c:428 Executing IVR menu auto_attendant_1 2011-08-30 13:13:46.164796 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16@8000hz 1 channels 30ms 2011-08-30 13:13:46.174819 [DEBUG] sofia.c:4761 Channel sofia/sipinterface_1/3335 entering state [ready][200] 2011-08-30 13:13:46.412814 [DEBUG] switch_rtp.c:3083 Correct ip/port confirmed. 2011-08-30 13:13:46.940791 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:47.360757 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:48.350702 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:48.390701 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:48.397689 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:48.872658 [DEBUG] switch_ivr_play_say.c:1648 done playing file 2011-08-30 13:13:48.872658 [DEBUG] switch_ivr_menu.c:343 waiting for 4/4 digits t/o 2000 2011-08-30 13:13:58.893140 [DEBUG] switch_ivr_menu.c:390 digits '' Packet capture shows this tidbit: p=Digit-Collection y=Digits d=7 I’m very new to this, any help or advice will be appreiciated. TIA, RP

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Re: [Freeswitch-users] Calls over 3G
by john＠mobex.biz 07 Sep '11

07 Sep '11

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Calls over 3G
by Wagner 07 Sep '11

07 Sep '11

Hello, What's the best way to make calls over 3G networks with less delay and more quality? any kind of compression, or different codec, any ideas? Thanks

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FreeSWITCH Conference Call Today
by Michael Collins 07 Sep '11

07 Sep '11

Hello all! We have a few items go over on today's conference call. I've been able to get caught up on the ChangeLog and, as usual, there are some new items that you may be interested in hearing about. Also, we could use some assistance in getting caught up in documenting all these things. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_09_07 Looking forward to talking to you soon! -Michael

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Re: [Freeswitch-users] - freeswitch bye issue
by Simon Leck 07 Sep '11

07 Sep '11

Hi Everyone, kindly help if you can At the moment I did encounter a issue, A leg user called a b leg user and when b leg user hanged up (Bye is not send by freeswitch to the A leg user) the only way I could hanged up this call to hanged up the A leg user manually? Would be great if someone could enlighten me on how this can be achieved? Thanks in advanced to anyone for helping me out Thanks again Simon Leck

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Connecting to FS from outside
by Yungwei Chen 07 Sep '11

07 Sep '11

Hi, I am trying to connect to FS from outside using XLite. In XLite settings, I set the domain to <the external IP address of FS>:5080 in order to use the external profile. The problem is that XLite is unable to register with FS due to the following error: 2011-09-06 09:32:52.030921 [WARNING] sofia_reg.c:2183 Can't find user [1000(a)71.197.220.29] You must define a domain called '71.197.220.29' in your directory and add a user with the id="1000" attribute and you must configure your device to use the proper domain in it's authentication credentials. In conf/vavrs.xml, domain is set to the local IP address of FS. How can I allow user 1000 to register with FS from outside? Thanks.

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