December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Cannot Kill stale channels
by Marc de Corny 20 Mar '13

20 Mar '13

Hi I have calls that come into a queue and then get pulled out by a lua script in background and transfered to a destination. all works fine except that it looks like the sessions a not clearing from FS when all the parties clear the calls. When I do show channels, I get this for example: uuid direction created created_epoch name state cid_name cid_num ip_addr dest presence_id presence_data callstate callee_name callee_num callee_direction call_uuid hostname sent_callee_name sent_callee_num 5e8215e7-b036-4cce-9b96-e57db92f13ee outbound 03/12/2012 09:32 1.35E+09 sofia/external/02031950164 CS_HANGUP Outbound Call 2031950164 135.196.144.32 agent_to_queue_paymentsense_queue_service ACTIVE SEND 5e8215e7-b036-4cce-9b96-e57db92f13ee freeswitch2 4.4209E+11 2089623100 But when I try to get kill or query it, I get a message that the call is not anywhere in the FS. /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_kill 5e8215e7-b036-4cce-9b96-e57db92f13ee" -ERR No Such Channel! /usr/local/freeswitch/bin/fs_cli -H 10.5.2.105 -x "uuid_exists 5e8215e7-b036-4cce-9b96-e57db92f13ee" false Does anyone have an idea how I can kill that call. If I restart the FS, they clear, but the problem is that they are hitting my limit on number of simultaneous calls. Any help much appreciated. Thansk marc

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Playing of mms stream by means of FS
by Yuriy Nasida 05 Mar '13

05 Mar '13

Hi guys What the best way for the playing of mms stream like this mms://live.rfn.ru/rcult_64 by means of FS ?I try to use mod_shout but without success. Do I have to use mplayer or vlc for this ?I saw this in wiki http://wiki.freeswitch.org/wiki/Mod_vlc But... probably there is the more better way ? Please adviseThanks.

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Re: [Freeswitch-users] Concatenated multiple playback types
by Peter Olsson 28 Feb '13

28 Feb '13

I don't believe this is possible. If you record 5 seconds of silence into a file you might be able to use file_string:// to play them both. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Emrah Skickat: den 22 augusti 2012 16:24 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Concatenated multiple playback types Hello there, I am trying to add some silence before the alone sound file in my conference.cnf.xml and can't get it to work. Here is what I tried: <param name="alone-sound" value="silence_stream://5000!file_string://conference/conf-alone.wav"/> <param name="alone-sound" value="silence_stream://5000&file_string://conference/conf-alone.wav"/> <param name="alone-sound" value="silence_stream://5000|file_string://conference/conf-alone.wav"/> <param name="alone-sound" value="silence_stream://5000;file_string://conference/conf-alone.wav"/> Any idea on how to do this? Thanks! Emrah _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:5034e9a632761303016055!

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ooH323 vs mod_h323
by Saeed Ahmed 17 Feb '13

17 Feb '13

Dear FSers, which one is better to use when there are higher number of concurrent calls? *ooH323 *or *mod_h323*. Thanks

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Nibblebill database update
by William Alianto 14 Feb '13

14 Feb '13

Hi, I was trying to use nibblebill as billing control of my FS. I have configured the odbc and the nibblebill configuration. When I tried to call, the query looks ok. But when I hanged up the call, it seems like there is something wrong with the database connection, since it's not updating the database at all. I got error message on CLI 2012-11-01 10:51:00.559472 [ERR] switch_odbc.c:494 ERR: [UPDATE accounts SET usage=usage-0.476000 WHERE user='1001'] [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'usage=usage-0.476000 WHERE user='1001'' at line 1 ] 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:343 ERR: [UPDATE accounts SET usage=usage-0.476000 WHERE user='1001'] [] 2012-11-01 10:51:00.559472 [CRIT] mod_nibblebill.c:542 Failed to log to database! 2012-11-01 10:51:00.559472 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT usage AS nibble_balance FROM accounts WHERE user='1001'] 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:385 ERR: [SELECT usage AS nibble_balance FROM accounts WHERE user='1001'] [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.5.24-0ubuntu0.12.04.1]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'usage AS nibble_balance FROM accounts WHERE user='1001'' at line 1 ] 2012-11-01 10:51:00.559472 [ERR] mod_nibblebill.c:386 Error running this query: [SELECT usage AS nibble_balance FROM accounts WHERE user='1001'] Could anybody help me solve this issue? Regards

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Specifying file permission for audio records.
by Mimiko 08 Feb '13

08 Feb '13

Hello. I want audio files recorded from callcenter to be available to a web server. Both webserver and FS runs on same server. The storage directory for holding recorded audio files is on the same server. FS run in unprivileged mode, and the directory for recorded audio files have permission 775. Webserver runs unprivileged under another user. The problem is that created recorded audio files by FS get the 770 permission. How to change this behavior, so permission of created files will inherit directorys permision? -- Mimiko desu.

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Log Rollover issue
by Nageshwara Rao Moova 07 Feb '13

07 Feb '13

Hi all, I have modified my default logconf file for rollover to be restricted to 100. But sometimes freeswitch is unable to rename rollover say “cannot rename freeswitch.log.41” and fails. But the fail follows with serious issue of writing all the logs to "freeswitch.log" and ends up filling the disk space. I have not change the default file size i.e 10MB. How does 10MB size overridden by freeswitch? -- regards & thanks -- m nageshwara rao 99891 86280

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NAT traversal - the final say..!
by Cal Leeming [Simplicity Media Ltd] 05 Feb '13

05 Feb '13

*Any and all feedback on this thread would be much welcomed.* Hello, * * There seems to be a large number of discussions surrounding NAT traversal, as well as lots of documentation, but with no concrete answers. The NAT related wiki documentation is tedious, and depending on the outcome of this thread, I'd like to spend some time cleaning it up. The most common problem (the same as ours) was having a router with broken ALG and a softphone that does not seem to work with STUN. The following REGISTER is sent from a phone. REGISTER sip:1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:57787 ;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:2000@192.168.1.102:57787;rinstance=0c7190b115a36513> To: "foxx"<sip:2000@1.2.3.4:5060> From: "foxx"<sip:2000@1.2.3.4:5060>;tag=83311448 Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI. CSeq: 7 REGISTER Expires: 120 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Supported: replaces User-Agent: 3CXPhone 6.0.25732.0 Content-Length: 0 As you can see, the client's public IP is not specified anywhere. FreeSWITCH offers several ways around this, the main ones being; * NDLB-connectile-dysfunction * NDLB-force-rport * apply-nat-acl * sip-force-contact The one that has worked in our case was "NDLB-connectile-dysfunction" (otherwise known as NAT HACK), however there seems to be a lot of negative comments about using this. >From what I can tell, the general argument is that NAT HACK is considered a non RFC compliant hack, and the SIP phones should be doing a better job of keeping to the RFCs. In principle, this is a fair argument - but in practise, it's not a reasonable assumption that all phones are RFC compliant, and (imho) not a reasonable argument to have this functionality disabled by default. So, I'd like to present the following arguments; * Are there any other negative aspects about using NDLB-connectile-dysfunction, other than it is a non compliant RFC hack? * Why is NDLB-connectile-dysfunction not enabled by default when certain conditions are met? In the event that FreeSWITCH receives a REGISTER from a phone specifying a Contact/Via as 192.168.0.0/16, but received on a public IP, then it should be obvious that NAT is broken and automatically try to circumvent it. * People seem to get confused between server side and client side NAT problems, and that they both need to be resolved in a different way. The documentation doesn't seem to reflect this clearly.

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Callgroup keeps ringing after answer - changed from shared extensions
by Mario G 04 Feb '13

04 Feb '13

Been at this for 3 weeks now, hope someone can help.... This started after replacing shared extensions with unique users and updating from an early 2011 git. Ring group via "pointer" works fine as it always has. However, answer via callgroup variable intermittently (30-40%) leaves one or mores phone ringing even though the the call was answered at another extension. The darn thing is that I can't reproduce it myself to get a trace. Any help/suggestions greatly appreciated. Phones are SPA962 (UDP) and Bria on iPad (TCP, same problems) and they all stopped ringing when they shared IDs. Mario G default.xml ring-group has (abbreviated to save space)------------------: <group name="home"> <users> <user id="100" type="pointer"/> <user id="101" type="pointer"/> ..... more users </users> </group> <group name="work"> <users> <user id="200" type="pointer"/> <user id="201" type="pointer"/> </users> </group> </groups> Extension----------------------: <user id="101"> User: <params> <param name="password" value="$${default_password}"/> </params> <variables> <variable name="voicemail_domain" value="xxxxxx.com"/> <variable name="toll_allow" value="domestic,domestic10,international,local"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Front Desk"/> <variable name="effective_caller_id_number" value="101"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>  <variable name="callgroup" value="home"/> </variables> </user> Bridge command-----------------: <extension name="xxphone"> <condition field="destination_number" expression="^2920$"> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>  <action application="execute_extension" data="2925 XML default"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ringback=file_string://${xxsounds}hi-long.wav!${sayname}!tone_stream://${us-ring};loops=-1"/>  <action application="bridge" data="{originate_timeout=17}${group_call(home@${domain_name}+A)}"/>

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mod_httapi problem when trying to play an audio hosted in S3 amazon cloud
by Leonardo N. S. Pereira 01 Feb '13

01 Feb '13

Hi. I'm having problem when trying to play an audio hosted in S3 Amazon cloud The MP3 URL is: / / /h//ttp://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2Fblind_willie.mp3?Expires=1349381157&AWSAccessKeyId=AKIAJKQT6QPSI3BXJJ4A&Signature=bnJgjbnxqlgsdYIcKIbYUa9gqao%3D/ The result is: 2012-10-04 20:27:28.891298 [DEBUG] switch_ivr.c:605 sofia/external/+15752222222(a)67.231.8.195 Command Execute playback(http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2… EXECUTE sofia/external/+15752222222(a)67.231.8.195 playback(http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2… 2012-10-04 20:27:28.903180 [ERR] mod_httapi.c:2545 File at url [http://test-media-dev.s3.amazonaws.com/u-asevyeard22pl6sc2z4knzi%2Fblind_wi…] is unreachable! If I use the Browser with same URL, I'm able to listen to the sample audio. Any idea about what is wrong? Thanks in advance, Leonardo

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