June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Too many channels used in my scenario
by Mino Haluz 11 Jun '13

11 Jun '13

Hi, I am thinking of this topology: caller->freeswitch->proxy in proxy, there will be some routing module as carrierroute or so and then call is sent back callee<-freeswitch<-proxy thing is, if only one call is placed, 4 channels are used. If I used sangoma transcoding card, every channel has to be transcoded to the internal FS coded and back, so 8 transcoding channels in sangoma driver are used for just one call (!). How could I tell freeswitch to do not pass SDP information to the proxy side and let it use only 2 channels as in standard caller->callee call. Thanks, Mino

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How to add username in SIP packet OPTIONS
by "Гришин В.С." 11 Jun '13

11 Jun '13

My gataway profile <include> <gateway name="SIP_M200"> <param name="username" value="USERNAME_STRING"/> <param name="realm" value="172.30.0.2"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="5"/> </gateway> </include> SIP packet which is accepted by the remote UA Thu Nov 29 16:38:24 2012 RECEIVE 15:5060 <-- 172.30.0.1:5080 OPTIONS sip:172.30.0.2;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.30.0.1:5080;rport;branch=z9hG4bK6DKm54N817tUS Max-Forwards: 70 *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ To: <sip:172.30.0.2>* Call-ID: 79818268-b4c4-1230-60ac-00d0b7ded66a CSeq: 36758947 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20120920T220849Z~f718a5e8e6 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 but in reply arrives nothing... the developer points to that that in a SIP packet there is no field username in *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ <----in this string no username To: <sip:172.30.0.2> <---- and in this string no username* it is necessary to make as From: "USERNAME_STRING" <sip:USERNAME_STRING@172.30.0.1>;tag=9DcB7gSNZZ41j What it is necessary to change or add in a gateway profile (or other place) for this purpose? Vladislav Grishin

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any updates for Mod_cdr?
by Vincent Xia 11 Jun '13

11 Jun '13

i see this at http://wiki.freeswitch.org/wiki/Mod_cdr - ODBC (Coming soon pending somebody being able to actually test it) is there any updates now? i was thinking of generating CDRs into sql server.

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how to set the recording behavior in session:recordFile()?
by Vincent Xia 11 Jun '13

11 Jun '13

is it possible to set the recording behavior(record a-leg, b-leg, or both) in session:recordFile(), like we do in the dialplan: <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M -%S)}.wav"/> in my point of view, session is a handle of one call leg, so by session:recordFile along it may not work out, any clue? BR

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FreeSWITCH Weekly News and Notes
by Michael Collins 10 Jun '13

10 Jun '13

Happy 2nd week of June to everyone. We want to remind everyone that on June 19th we will be releasing the ClueCon 2013 <http://www.cluecon.com> schedule on the FreeSWITCH community conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_19>, so please be there. As an added bonus the whole FreeSWITCH dev team will be on hand and hanging out with the community. On occasion we like to acknowledge community members who've stepped up and made a difference in a way that benefits the FreeSWITCH project as a whole as well as those who use it. Our friend Call Leeming has been doing a great job of doing triage on the FreeSWITCH bug tracker<http://jira.freeswitch.org>. In fact, just today he reported<http://lists.freeswitch.org/pipermail/freeswitch-users/2013-June/096500.html>on checking five Jira tickets. We appreciate these efforts and we encourage everyone to take a few moments to peruse the open tickets and look for items where they can assist, even if it's merely testing the validity of a bug report. Thank you to everyone who helps out. After having open community discussions these past few weeks we will have a formal presentation this week<http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_12>by yours truly. In response to numerous requests we have decided to do a demonstration of the CudaTel <http://www.cudatel.com> Communication Server. I will be giving a tour of the CudaTel user interface and we will be glad to answer questions. I think you will be impressed with the hard work that the core FreeSWITCH team has put into building the CudaTel. We look forward to talking to you on Wednesday. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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ICTDIALER stays on 300 concurrent calls + some log files
by Sayyed Mohammad Emami Razavi 10 Jun '13

10 Jun '13

BTW FS cpu usage is 85% with this command "ps -eo pcpu,pid,user,args | sort -k 1 -r | head -20" and returns: %CPU PID USER COMMAND 86.9 1871 root /usr/local/freeswitch/bin/freeswitch -nc 7.6 1929 root /usr/local/plivo/bin/python /usr/local/plivo/bin/plivo-rest -d -c /usr/local/plivo/etc/plivo/default.conf -p /usr/local/plivo/tmp/rest-default.pid 41.8 1672 mysql /usr/libexec/mysqld --basedir=/usr --datadir=/var/lib/mysql --user=mysql --log-error=/var/log/mysqld.log --pid-file=/var/run/mysqld/mysqld.pid --socket=/var/lib/mysql/mysql.sock 3.7 4467 atel /usr/bin/php -q /var/www/ictdialer/www/modules/broadcast/broadcast_service 668 1060 3.5 4469 atel /usr/sbin/httpd 3.5 4461 atel /usr/sbin/httpd 3.5 4458 atel /usr/sbin/httpd 3.4 7817 atel /usr/sbin/httpd 3.4 4470 atel /usr/sbin/httpd 3.4 4456 atel /usr/sbin/httpd 3.4 4455 atel /usr/sbin/httpd 3.3 4464 atel /usr/sbin/httpd 3.3 4459 atel /usr/sbin/httpd 3.3 18698 atel /usr/sbin/httpd 3.2 4462 atel /usr/sbin/httpd 3.2 4460 atel /usr/sbin/httpd 3.1 4457 atel /usr/sbin/httpd 18.7 1938 root /usr/local/plivo/bin/python /usr/local/plivo/bin/plivo-outbound -d -c /usr/local/plivo/etc/plivo/default.conf -p /usr/local/plivo/tmp/outbound-default.pid 1.0 13471 root ps -eo pcpu,pid,user,args fs version is 1.5 fs_cli> status freeswitch@internal> status UP 0 years, 0 days, 1 hour, 12 minutes, 42 seconds, 448 milliseconds, 342 microseconds FreeSWITCH (Version 1.5.0 ) is ready 14668 session(s) since startup 249 session(s) - 4 out of max 100 per sec 1288 session(s) max min idle cpu 0.00/67.00 Current Stack Size/Max 240K/8192K ---------- Forwarded message ---------- From: Sayyed Mohammad Emami Razavi <emamirazavi(a)gmail.com> Date: Mon, Jun 10, 2013 at 9:28 PM Subject: ICTDIALER stays on 300 concurrent calls To: freeswitch-users(a)lists.freeswitch.org ICTDIALER makes calls from 1 to 300 but it stays on almost 300 connection. I have 1200 ports(40 E1) but it can not exceed 300 concurrent calls! any configuration should i set in FS, Plivo, Httpd or ...? i have HP G8 with ICTDIALER VM 8 cores and 16GB RAM! any suggestion guys? PLZ help me asap! regards.

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ICTDIALER stays on 300 concurrent calls
by Sayyed Mohammad Emami Razavi 10 Jun '13

10 Jun '13

ICTDIALER makes calls from 1 to 300 but it stays on almost 300 connection. I have 1200 ports(40 E1) but it can not exceed 300 concurrent calls! any configuration should i set in FS, Plivo, Httpd or ...? i have HP G8 with ICTDIALER VM 8 cores and 16GB RAM! any suggestion guys? PLZ help me asap! regards.

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OSX 10.8.4 update note
by Mario M Guzman 10 Jun '13

10 Jun '13

For interested parties: Built FS 1.5.2 from scratch on OSX 10.8.4 and all works as before. (Fine except for fs-5375 and fs-5223). Wiki is updated. Mario G

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DTMF rfc2833 hanging up the calls
by Mino Haluz 10 Jun '13

10 Jun '13

Hi, if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call. n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL] 2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK] 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint? Thanks, Mino

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2013-02-15 05:30:00.934663 [DEBUG] mod_skypopen.c:1056 [e5e3e4b|3d6ed64] [DEBUG_SKYPE 1056 ][gplhost_skype1 ][DIALING, INC_HNG] NO SPACE in WRITE BUFFER: there was no space for 640
by Shawndell Harris 10 Jun '13

10 Jun '13

Placing a call out through skype i receive this error 2013-02-15 05:30:00.934663 [DEBUG] mod_skypopen.c:1056 [e5e3e4b|3d6ed64] [DEBUG_SKYPE 1056 ][gplhost_skype1 ][DIALING,INC_HNG] NO SPACE in WRITE BUFFER: there was no space for 640 -- Shawndell Harris

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