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Debian package dependency:
by Julian Pawlowski 11 Jun '13

11 Jun '13

Hi, since a few days the package dependencies for freeswitch-meta-codecs seems to be broken in the 1.2 stable tree. I get the following message during install: The following packages have unmet dependencies: freeswitch-meta-codecs : Depends: freeswitch-mod-b64 (= 1.2.11~n20130608T103852Z-1~wheezy+1) but it is not installable I had a look to our Debian repository and yes, the named package freeswitch-mod-b64 indeed is missing. However it exists in the 1.3 unstable tree so I guess it's new and the reference to this package was mistakenly transported to the stable tree. Travis, would you mind to correct this so our Gemeinschaft5 build will be running again? ;-) That would be awesome! Cheers, Julian

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Re: [Freeswitch-users] NAT problem: Use local IPv4 for specific LAN IP address range
by Ben 11 Jun '13

11 Jun '13

Steve, Sorry for the late reply (4 months!!!), the immediate need for a fix went away, until now. The problem with setting up an additional profile would be that the end user would have to change port numbers when moving from the internal network to an external network. Unfortunately the end user does not have access to these sorts of settings with the app we're using, and more importantly, I need this to be seamless between networks. Does anyone else have any ideas? I'm starting to think this problem is unsolvable with the current functionality. Implementing an equivalent of the Asterisk "nat.conf" with a "localnet=" line, which will force the Freeswitch local IPv4 address as the SDP for any requests originating from one of these "localnets" would hopefully resolve the issue. Ben From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Steven Ayre Sent: 07 February 2013 13:07 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Use local IPv4 for specific LAN IP address range One simple option would be to have a 2nd profile listening on another port (eg 5080) that does not use ext-*-ip and connect to that internally. -Steve On 7 February 2013 11:44, Ben <ben122uk(a)gmail.com> wrote: Greetings all, A question on NATing in Freeswitch - I have my internal SIP profile to use a static public IP for the ext-sip-ip and ext-rtp-ip. The Freeswitch server is on a local LAN IP, for example 192.168.0.10. External SIP clients can access the server via the external IP, as there's a static NAT on the router to map the public IP to the private IP. I have another SIP client on a different LAN, but behind the same firewall, for example, 10.0.0.10. This client is allowed to talk to the server on its private ip of 192.168.0.10, by allowing inter-vlan traffic to pass through. My challenge is, how do I get Freeswitch to recognise this internal SIP client as being allowed to use the internal IP for SIP/RTP communications, and therefore present the client with internal addresses in the SIP header for SIP/RTP? Due to statically setting ext-sip-ip and ext-rtp-ip, Freeswitch tells the client to go via the public IP. The firewall has a policy to not accept traffic to the public IP if sourced from an internal device. I have explored the NATing options in Freeswitch, but can't find anything that fits my scenario. It seems to be more towards how to enable NAT in certain situations, not disable it. I have also looked at re-writing the SDP, but this re-writes the incoming SDP from the clients. My intention was to re-write the outgoing SDP from the Freeswitch to include the internal IP address as the RTP connect in address. I don't have much Asterisk experience, but I seem to remember a nat.conf file where you could specify subnets that were treated differently. These could be classed as internal subnets, and would be given the private IP in the SDP from asterisk, everything got the public IP. I'm also aware that I could create another SIP profile easily with different NATing/IP options, but I want to find out if there's anything I can do with the above first. Many thanks for the help! _________________________________________________________________________ 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

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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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