December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch
by madan.mallikarjun 18 Dec '12

18 Dec '12

Hello All, I m new to freeswitch and need all the help from where to start and how to proceed . Is there any sort of documented way how FS can be installed coz frankly freeswitch seems to be scattered. thanks Madan -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-tp7585608.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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G729 - [INCOMPATIBLE_DESTINATION]
by Balamurugan Mahendran 18 Dec '12

18 Dec '12

Need help on Codec Negotiation 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare * [G729:0:8000:20:64000]/[G729:18:8000:20:8000]* 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[SILK:117:8000:20:20000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[SILK:118:12000:20:25000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[SILK:119:16000:20:30000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[SILK:120:24000:20:40000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:0:8000:20:64000]/[GSM:3:8000:20:13200] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:120:24000:20:40000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-12-16 11:35:52.246422 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf send/recv payload to 101 2012-12-16 11:35:52.246422 [DEBUG] switch_channel.c:2852 (sofia/internal/ 501(a)50.54.12.39) Callstate Change DOWN -> HANGUP 2012-12-16 11:35:52.246422 [NOTICE] sofia.c:5781 Hangup sofia/internal/ 501(a)50.54.12.39 [CS_NEW] *[INCOMPATIBLE_DESTINATION]* Thanks, Bala

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FreeSWITCH Weekly News and Notes
by Michael Collins 18 Dec '12

18 Dec '12

A happy and cold (in the northern hemisphere) Monday to you all! I'd like to start this week's news and notes by alerting everyone to the fact that there's been a significant addition to the functionality of the XML dialplan. If you've been around FreeSWITCH for any length of time you've probably read the words, "Nested conditions not allowed!" in yours or someone else's FreeSWITCH logs. As of last Friday<http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=9b7b46dbca9c4c458…>that has changed! In response to Jira FS-4935 <http://jira.freeswitch.org/browse/FS-4935> Anthony has added provisional support for nested <condition> tags inside the dialplan. (Thanks to IRC user vipkilla for adding this to the wiki<http://wiki.freeswitch.org/wiki/Dialplan_XML#Nested_Conditions>already.) As you may know we are working the second edition of the FreeSWITCH "bridge" book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tel…>. I will be updating chapters 5 and 8 to reflect this new change. It seems appropriate that with this new feature we should talk about it on this week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_19>. Ken Rice and I will work up some simple examples of how to use the nested conditions and how they relate to the existing XML dialplan controls such as the break attribute and the regex tag. If you have a dialplan example that works well with nested conditions please email Ken and me off list. One last reminder for our Windows users: Dave Kompel shared with us some useful information <http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12> for gathering debug data when FreeSWITCH crashes under Windows. Our other main Windows guru, Jeff Lenk, was also on the call and gave some helpful input. If you are running under Windows you now have more tools at your disposal with which to analyze crash data and open Jira tickets. Have a great week and we'll talk to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Linksys Registration disappears after 5 minutes
by Sean Devoy 18 Dec '12

18 Dec '12

Hi, I have a Linksys ATA (RPT300). I have it successfully Registering through a NAT connection. But, 5 minutes later FS no longer shows it registered. The RTP300 still says it is registered and can of course dial out, but not in. For what it is worth, this RTP300 has been working for months and months. Since we now have Polycom phones on this LAN, we had to enable SIP ALG and NDLB. Cisco phones and Polycom phones are all working great now, but I cannot figure out what setting to change in the Linksys device. There are no words to express how much I hate NAT and SIP. Thanks, Sean

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Build error with mod_spandsp
by Nikolay Rogoshchenkov 18 Dec '12

18 Dec '12

Hello, Are anybody know how can we solve problem with building *mod_spandsp* for FreeBSD 8.2. I also also tried "*gmake spandsp-reconf*" (as recommended in FS-4577<http://jira.freeswitch.org/browse/FS-4577>), but results are the same: [...] *making all mod_spandsp* *Making all in src* *libtool: compile: gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/local/include -I/usr/local/include/libxml2 -I/usr/src/freeswitch/libs/tiff-4.0.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DHAVE_VISIBILITY=1 -g -O2 -MT t4_rx.lo -MD -MP -MF .deps/t4_rx.Tpo -c t4_rx.c -fPIC -DPIC -o t4_rx.o* *t4_rx.c:85: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'tiff_fx_field_array'* *t4_rx.c: In function 'write_tiff_image':* *t4_rx.c:320: warning: implicit declaration of function 'TIFFCreateCustomDirectory'* *t4_rx.c:320: error: 'tiff_fx_field_array' undeclared (first use in this function)* *t4_rx.c:320: error: (Each undeclared identifier is reported only once* *t4_rx.c:320: error: for each function it appears in.)* *t4_rx.c:327: warning: implicit declaration of function 'TIFFWriteCustomDirectory'* *gmake[7]: *** [t4_rx.lo] Error 1* *gmake[6]: *** [all] Error 2* *gmake[5]: *** [all-recursive] Error 1* *gmake[4]: *** [/usr/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2* *gmake[3]: *** [mod_spandsp-all] Error 1* *gmake[2]: *** [all-recursive] Error 1* *gmake[1]: *** [all-recursive] Error 1* *gmake: *** [all] Error 2* *#* Thank you! -- nikk

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Video call
by Emrah 18 Dec '12

18 Dec '12

Hi there, I am having a hard time getting video working. I have mod_h26x loaded. I tried setting proxy_media=true before bridging. I tried forcing codecs with absolute_codec_string. I tried inherit_codec=true. I get no video no matter what I do. The codec I need is H264. What is the procedure to get video calling working on FS? Cheers and thanks, Emrah

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callcenter with external agents
by Juan Pablo L. 17 Dec '12

17 Dec '12

Hi, i m trying to set up a callcenter but the lines for the agents are mobile phones in a mobile network and are not attached to freeswitch, i m trying to set it up with the following: <agent name="1000@default" type="callback" contact="[call_timeout=10]sofia/external-huawei_gw/XXXXXX" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> XXXXX being the mobile number but i m getting the following error: Member 2018 <2018> in queue 'callcenter@default' reached max wait of 0 sec. with no agent plus join grace period of 5 sec. and i see that freeswitch does not even try to dial out to the trunk. i m wondering if configuring it the way i need it is even possible as i think the problem may reside in the fact that freeswitch does not actually see the agent registered. thanks!!

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Stereo recording of conference - audio on the two channels is out of sync
by Dave Horton 17 Dec '12

17 Dec '12

I am trying to make a stereo recording of a two-party bridged conference -- i.e. caller calls in, we create a bridged conference and outdial, and then get a stereo recording with caller and called party on different channels. My dialplan is shown below. The problem is that the two channels on the resulting recording are completely out of sync with each other. Does anyone have an idea about why this might be happening? (In case anyone is wondering why I am recording a 2 party conference, rather than just a bridged outdial, the reason is that I need to play an audio clip to both parties that the call is being recording, and I need that clip to also be captured in the recording, and I can't figure out a way to do that other than to create a conference and play the clip into the conference. I'm open to other suggestions though if anyone has an alternative approach). (Note: the lua script in the dialplan below just collects a number to outdial) <include> <extension name="617xxxxxxx" continue="false"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="617xxxxx"> <action application="ring_ready"/> <action application="answer"/> <action application="start_dtmf"/> <action application="set" data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/> <action application="export" data="outfile=$${base_dir}/tmp/${call_id}.wav"/> <action application="lua" data="demo_call2.lua"/> <action application="export" data="RECORD_STEREO=true"/> <action application="export" data="execute_on_media=record_session ${outfile}"/> <action application="export" data="api_on_answer=conference ${call_id} play misc/this-call-may-be-recorded.wav async"/> <action application="conference" data="bridge: ${call_id}@simple:{origination_caller_id_number=${caller_id_number}}sofia/normal_customer/${outdial}@${egress_gateway}"/> </condition> </extension> </include>

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ODBC error
by Archana Venugopan 17 Dec '12

17 Dec '12

Hi, Can anyone please tell me why I am getting this error inspite 'isql -v smepbx smepbx smeswitch' was working. [CRIT] switch_core_sqldb.c:433 Failure! ODBC NOT AVAILABLE! Can't connect to DSN smepbx Regards, Archana.V

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latest git version error
by Archana Venugopan 17 Dec '12

17 Dec '12

Hi, When I try to fetch latest git version am facing with the below error and am not able to get the git. From where can I get freeswitch latest version? [root@VECTONE-CLOUDE src]# git clone git://git.freeswitch.org/freeswitch.git Initialized empty Git repository in /usr/local/src/freeswitch/.git/ git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out git.freeswitch.org[0: 2606:d900:0:24:1024:ff:fe00:1234]: errno=Network is unreachable fatal: unable to connect a socket (Network is unreachable) Regards, Archana.V

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