November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH - Polycom and DNS SRV
by Lloyd Aloysius 03 Nov '11

03 Nov '11

Hi , I am testing the Polycom Phones [IP335,IP450 and IP6000] and DNS SRV records. All my DNS SRV records and other phone models [Cisico , Asstra, Yealink] working without any issue. But for Polycom Phones .. I need to specify the "Outbound Proxy" as IP address otherwise phone never register. * voIpProt.SIP.outboundProxy.address="IP ADDRESS"* I could not find why we need to specify "Outbound Proxy? How to avoid using IP address? Does Polycom phones fully support DNS SRV records? Any help is appreciated. Thanks and regards, Lloyd

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securing sip channel
by Tamer Higazi 03 Nov '11

03 Nov '11

Hi! I am interested to know if there is a way to secure the registration against account hacking. for example, I would create an SIP Account server side, there would be the possibility that only users with the static IP-Address could login to the assigned account. Is there one other way like kerberos ticket authentication or public secret key with ipsec?! For any advise I would thank you Tamer

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FW: Send Event/User Channel Variable from FreeSWITCH to RTMP client
by RaviRaj Mulasa 03 Nov '11

03 Nov '11

Hi Please let us know how to send an Event/ User Channel Variable from FreeSWITCH to a RTMP client. Thanks RaviRaj.

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How to paste text from Linux to pastebin.freeswitch.org?
by Yehavi Bourvine 03 Nov '11

03 Nov '11

Hello, I have some logfile on Linux and would like to copy it to pastebin. Is there some way to upload the file? Currently I mark each screen and copy it, but there must be an easier and more reliable way... Furthermore, this way all the escape sequences in the log are shown as "ESC" and not parsed. Thanks! __Yehavi:

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Call quality
by Hynek Cihlar 03 Nov '11

03 Nov '11

Would anybody know, if the call quality could be dependant on the number of media switches the call is routed through? Currently I'm testing a system with several freeswitches in the call way (up to 4) and I'm noticing very short gaps, not very disturbing but noticable. What's your experience? Sent from my mobile device

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Radius client problems
by Miha Zoubek 03 Nov '11

03 Nov '11

Hi, As now my freeswitch is working properly I would like to implement radius client on it for AAA. I have done as is written on http://wiki.freeswitch.org/wiki/Mod_rad_auth. My dialplan look like this: <extension name="test12345"> <condition field="destination_number" expression="^(10[03][0-9])$"> <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="USERNAME=1001"/> <action inline="true" application="set" data="PASSWD=1111"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="export" data="dialed_extension=$1"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> After I try to make phone call I get his: *2011-11-03 13:33:54.582920 [ERR] mod_rad_auth.c:347 Error loading radius dictionary 2011-11-03 13:33:54.582920 [ERR] mod_rad_auth.c:546 An error occured during RADIUS Authentication(RC=-1) 2011-11-03 13:33:54.582920 [ERR] mod_rad_auth.c:702 An error occured during radius authorization.* ok, I thought that link location is wrong but it is not: <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/ cd /usr/local/etc/radiusclient/ [root@localhost radiusclient]# ls dictionary dictionary.cisco dictionary.merit issue radiusclient.conf dictionary.ascend dictionary.compat dictionary.sip port-id-map servers [root@localhost radiusclient]# If I put location like this /usr/local/etc/radiusclient/dictionary, the dictionary is found (but only one) but I get error that id-conf-h323 is missing (I put it in dictionary). My questions: Why I am getting first error if location is right? What are second two errors about? Thank you for all your help! BR, Miha

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IM isn't working
by Nicolás Menoni 03 Nov '11

03 Nov '11

Hello, I'm having a problem that maybe at first it sounds basic but I couldn't solve it yet. The IM (instant message) was working for a time but after a moment it didn't work more. I'm able to send IMs from FS_CLI using "chat" command, but I can't make that the sip users could chat between them. The FS_CLI doesn't give me any info about the problem, I can't see anything from the logs, it llooks like as if everything is ok. Any help? Thank you!

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Re: [Freeswitch-users] SOLVED Public IP in Contact header
by Hynek Cihlar 03 Nov '11

03 Nov '11

I played with the ACLs but it didn't have any effect on the contact header. So I went into the code and after some investigation I stepped on the * sip-force-contact* variable. When this guy is set to * NDLB-connectile-dysfunction*, freeswitch overwrites the contact header with the IP/PORT it received the IP packet on, regardless of the actual nat state. Hynek On Wed, Nov 2, 2011 at 6:42 PM, Hynek Cihlar <hynek.cihlar(a)gmail.com> wrote: > Brad, Brian, thanks for your replies > > Actually, the switch is not used in the standard setup, where the external > profile is used as a gateway and the internal (default) profile for > directly connected sip clients. > > In my case, the switch has two profiles. The so called external is > connected to another switch, just passing calls. The connection is routed. > The second profile (internal) which is giving me troubles, is used for > registering SIP clients. The SIP clients are usually behind their own NAT > routers. > > SIP-IP and EXT-SIP-IP are both the same public IP, since the router is > publicly visible on the internal profile. On the external profile, they are > assigned to a private internal IP. The switch is sitting on two subnets. > > Now, the local-network-acl is set to localnet.auto on both profiles. I was > assuming ACLs are used solely for security! > > Hynek > > > > On Wed, Nov 2, 2011 at 6:16 PM, Brian West <brian(a)freeswitch.org> wrote: > >> You also have to put in local-network-acl so it knows when to lie and >> when not to lie in the contact. >> >> /b >> >> On Nov 2, 2011, at 11:54 AM, Brad Mina wrote: >> >> RTP-IP 178.238.X.Y >> Ext-RTP-IP 178.238.X.Y >> SIP-IP 178.238.X.Y >> Ext-SIP-IP 178.238.X.Y >> >> Let's take a look at what 'ext' might be! Keep in mind you'll need to >> change it on your external profile as well. >> >> >> >> >> 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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Send Event from FreeSWITCH to RTMP client
by RaviRaj Mulasa 03 Nov '11

03 Nov '11

Hi Please let me know how to send an Event from FreeSWITCH to a RTMP client. I saw a Java script function in index.html in Flex Client provided by FreeSWITCH : function onEvent(data) { onDebug("Got event: " + data); } originate rtmp/rtmp-session-id &event(TEST) , no luck! Thanks RaviRaj.

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mod_rtmp login
by rsaavedra＠ecogizmos.com 02 Nov '11

02 Nov '11

Hello, I'm trying to use mod_rtmp but everytime I tried on the console I see User not authorized. I was trying with user:1000 and password:1234. In the file rtmp.conf.xml I have all by default: <param name="bind-address" value="0.0.0.0:1935" /> <param name="context" value="public" /> <param name="dialplan" value="XML" /> <param name="auth-calls" value="true" /> <param name="buffer-len" value="50" /> <param name="chunksize" value="512" /> Best regards, Ricardo Saavedra

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