July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

RTP NAT problem in Freeswitch 1.2.3
by Nuno Reis 12 Jul '13

12 Jul '13

Good day all. I'm experiencing the following sinptom when using some softphones behind nat on a private LAN, sometimes the same happen with hardphones. Here's the scenario: FS : <public IP> _______ <public IP>LAN ROUTER <private LAN> --- softphone Basically when softphone makes an INVITE to FS it always sends the private IP on the SDP and when the media flow starts it's being sent out by FS to the public lan address resulting on a audioless call. However if the phone sends the public IP on the SDP there's no issue at all. I know there's a variable available disable_rtp_auto_adjust that shoud make freeswitch ignore the SDP IP and use the INVITE IP instead, but it isn't working for me. Here's what i currently have on my internal SIP profile: <profile name="internal"> <aliases> </aliases> <gateways> </gateways> <domains> <domain name="all" alias="true" parse="false"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="H264,G722,PCMA,GSM"/> <param name="outbound-codec-prefs" value="H264,G722,PCMA,GSM"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="<PUBLIC_IP>"/> <param name="sip-ip" value="<PUBLIC_IP>"/> <param name="hold-music" value="local_stream://moh"/> <param name="apply-inbound-acl" value="domains"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="local-network-acl" value="localnet.auto"/> <param name="record-path" value="/opt/freeswitch/recordings"/> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="presence-privacy" value=""/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="true"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="/opt/freeswitch/conf/ssl"/> <param name="tls-passphrase" value=""/> <param name="tls-verify-date" value="true"/> <param name="tls-verify-policy" value="none"/> <param name="tls-verify-depth" value="2"/> <param name="tls-verify-in-subjects" value=""/> <param name="tls-version" value="sslv23"/> <param name="odbc-dsn" value="freeswitch:user:password"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_from"/> <param name="ext-rtp-ip" value="<PUBLIC_IP>"/> <param name="ext-sip-ip" value="<PUBLIC_IP>"/> <param name="presence-hosts" value="_DISABLED_"/> <param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="NDLB-broken-auth-hash" value="true"/> <param name="dbname" value="share_presence"/> <param name="send-presence-on-register" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="registration-thread-frequency" value="30"/> <param name="enable-timer" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="send-message-query-on-register" value="true"/> <param name="all-reg-options-ping" value="true"/> <param name="sip-force-expires" value="3600"/> <param name="sip-expires-max-deviation" value="300"/> <param name="multiple-registrations" value="contact"/> </settings> </profile> Any suggestions on how to make FS use the INVITE IP for RTP instead of using the IP on the SDP? Looking forward to hear from you. Best Regards, * Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…>| www.wavecom.pt** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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ClueCon 2013 Reminder: Register Today to Maximize Your Chances of Winning
by Michael Collins 12 Jul '13

12 Jul '13

This is just a friendly reminder that if you register for ClueCon 2013<http://www.cluecon.com>by the end of today you will have the best chance of winning some of our great prizes. We have Android tablets, SIP phones, paperback and e-books, and more! We can't wait - only 24 days until ClueCon! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Bypass media when connecting two incoming calls
by Oleg Stolyar 12 Jul '13

12 Jul '13

Hi guys, Is there a way to bypass the media when connecting two incoming calls? For example Caller A calls in and is put into park (or valet park) Then Caller B calls in and is connected to Caller A. Is there a way to enable bypass media in this case or does it only work for a bridge? Thank you *Oleg*

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mod_skypopen skype client stability
by Arsen 12 Jul '13

12 Jul '13

Hi Freeswitchers, Does somebody have experience with running multiple skype interfaces with the same user id and skype version 4.x? Is it stable? We faced a skype crash issue under CentOS 6.4 and skype 2.0.0.72 static oss (provided by interactive installer), so we switched to Ubuntu 12.10 with improved memory and CPU configuration, but unfortunatelly skype interfaces keep crashing unexpectedly. We are using shell script which checks failed skype interfaces and remove them (sk remove) from the skypopen, then start the skype and sk reload, but sometimes it causes freeswitch to segfault :/ Any ideas are very welcome. FreeSWITCH Version 1.5.3b+git~20130705T151245Z~9202f0398f (git 9202f03 2013-07-05 15:12:45Z) skype 2.0.0.72 oss Thanks, Arsen.

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Re: [Freeswitch-users] how to configure pin for use in gsmopen ?
by Luis Azedo 12 Jul '13

12 Jul '13

Hi Giovanni, it works great! tried in windows and linux (linux guest on windows virtualbox host with usb passthru) in windows is very to configure since you can check which ports are used in device manager. in linux is more try and error , when we connect the usb it creates 3 ttyUSB* devices. thanks > > ---------- Forwarded message ---------- > From: Giovanni Maruzzelli <gmaruzz(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 5 Jul 2013 11:49:21 +0200 > Subject: Re: [Freeswitch-users] how to configure pin for use in gsmopen ? > Hello Luis, > > seems your mod_gsmopen cannot open the ttyUSB ports. > > This can be because those ports are not accessible from the user > FreeSWITCH is running as. Try to run FreeSWITCH as root, and see if it > works, or chmod the ttyUSB* to be rw for FreeSWITCH's user. > > Or, it can be because those ports are not in a usable state, because > were not initialized by Linux. Dongles have a complex machinery > bringing them from a USB-disk state to a USB-modem state. > > I believe you must at least install usb_modeswitch, and all the udev > machinery. I'm not sure those are installed in a minimal centos > installation. > > So, install the usb_modeswitch package and all related stuff (eg: > udev, etc). Or, you can try to compile usb_modeswitch and use it at > boot in an /etc/rc.local. usb_modeswitch very useful site is: > http://www.draisberghof.de/usb_modeswitch/ > > Good luck, and let us know. > > -giovanni > > >

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does FS support any kind of MMS?
by Vincent Xia 12 Jul '13

12 Jul '13

does FS support any kind of MMS SMS is ok, but what about MMS, multimedia messaging service?

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issues on mod_sms to confirm
by Vincent Xia 12 Jul '13

12 Jul '13

issues on mod_sms to confirm there are some issues on mod_sms i need to confirm: 1. mod_sms supports sending message in pager/message mode, but not session mode, neither MSRP is supported, right? 2. mod_sms does not provide any record for sms sent/replied between user agents as it does(CDR) for audio calls, right?

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Nagios Monitoring
by John Kennedy 12 Jul '13

12 Jul '13

Hello all, Are there any Nagios modules that you all can recommend for Freeswitch? I have not found too many and want to get peoples opinions about the few that do exist. What do people use to monitor their Freeswitch installations? Thanks, John John Kennedy If you always do what you've always done you'll always be where you've always been - Unknown (if you can show me who said this, please do) It is impossible to defeat an ignorant man in argument - William G McAdoo I am not an advocate for frequent changes in laws and Constitutions. *But laws and institutions must go hand in hand with the progress of the human mind.* As that becomes more developed, more enlightened, as new discoveries are made, new truths discovered and manners and opinions change, with the change of circumstances, institutions must advance also to keep pace with the times. We might as well require a man to wear still the coat which fitted him when a boy as civilized society to remain ever under the regimen of their barbarous ancestors. -- Thomas Jefferson

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Realtime Call stats
by adahary 11 Jul '13

11 Jul '13

Hi, I'm bridging calls between two FS boxes (different region locations) where both are setup with proxy-media (RTP passthru). I would like to measure calls performence in real time. How to collect FS realtime RTP call stats of: - Packet loss - Jitter - Round trip delay. thanks Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Realtime-Call-stats-tp7592766… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Limiting available codecs in bypass media mode
by Phil Quesinberry 11 Jul '13

11 Jul '13

I'm wondering what the best way is to limit the available codecs when making outbound bypass-media calls to the PSTN. The scenario is as follows: Extension phones have HD codecs including G722 enabled for the best audio quality between extensions for internal calls. When an outbound call is made to the PSTN, the first-choice wholesale provider we use offers (and seems to prefer) G722. We'd rather ensure that G711 is used instead. Despite the fact that G722 is not enabled for any of the codec prefs (I grep'ed the entire conf dir path to be sure), FS still passes along the offer of G722 from the endpoint, presumably since it's operating in bypass-media mode. Is there a setting to limit the codecs despite having bypass media set? Could we work around this by getting rid of the global bypass media setting and then setting bypass-media before bridging the call? Thanks, Phil Quesinberry Q Systems Engineering, Inc. Embedded Hardware/Software Development and VoIP Business Telephone Hosting Improve your business telephone services and save money (410) 969-8002 http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/>

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