July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Help
by Sanja Angelova 25 Jul '11

25 Jul '11

Dear all! I am new to FreeSwitch and I am trying to map a number for example 654321 to sip adress like this 1005(a)10.1.1.1 with configuring a local DNS server. When I write enum 654321 in fs_cli it gives me: Offered Routes: Order Pref Service Route ============================================================================== 10 100 E2U+sip sofia/${use_profile}/1005(a)10.1.1.1 and seems to be fine. But when I make a call from my soft phone, the FreeSwitch ends up my call and it gives me error 503 service unavailable. I have no idea what the problem is, can you please help me? Thank you in advance

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Re: [Freeswitch-users] One way calling issue over an internal trunk
by Mitch Johnson 25 Jul '11

25 Jul '11

Concerning the NAT. I will always be on the internal network, all external phones will come in on a VPN and then connect onto FreeSWITCH using the VPN. I did the ./freeswitch -nonat. I also changed the sip-ip and rtp-ip to the internal ip of the Freeswitch server on both external and internal xml files: Internal: <param name="rtp-ip" value="172.16.200.60"/> <param name="sip-ip" value="172.16.200.60"/> external: <param name="rtp-ip" value="172.16.200.60"/> <param name="sip-ip" value="172.16.200.60"/> <param name="ext-rtp-ip" value="local-network-acl"/> <param name="ext-sip-ip" value="local-network-acl"/> I'm still plodding around FreeSWITCH so if I did this wrong please let me know. Thanks, Mitch On Jul 25, 2011, at 6:43 AM, freeswitch-users-request(a)lists.freeswitch.org wrote: > From: Steven Ayre <steveayre(a)gmail.com> > Subject: Re: [Freeswitch-users] One way calling issue over an internal trunk > Date: July 25, 2011 3:05:39 AM EDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > I'll always be on the internal network so should never need the nat. > > Try starting FS with the -nonat option. It'll start faster and skip all the stuff where it tries to find your external IP and open routes through the NAT firewall. > > On your sip profiles set sip-ip and rtp-ip to your internal IP. > > Once you've done that you should have 2-way audio to anywhere on your LAN. > > -Steve > > > > > On 25 July 2011 05:08, Mitch Johnson <mitch.johnson7(a)gmail.com> wrote: > Sorry for the delayed response, got tied up in other things. A quick recap. I can dial from my call manager to Freeswitch but cannot dial from freeswitch to call manager. > > I wasn't sure what you meant by setting the ext-sip-ip and ext-rtp-ip. So I replaced the auto-nat with local-network-acl. I'll always be on the internal network so should never need the nat. > > I saw in the acl.conf.xml that it automatically populates all my local subnets. The networks are correct under the domains of the acl.conf. > > I'm not sure where else to look. Any help would be greatly appreciated. > > Thanks so much, > > Mitch > > On Jul 19, 2011, at 11:29 PM, freeswitch-users-request(a)lists.freeswitch.org wrote: > >> From: Brian West <brian(a)freeswitch.org> >> Subject: Re: [Freeswitch-users] One way calling issue over an internal trunk >> Date: July 19, 2011 8:55:02 PM EDT >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> >> >> you need to set the ext-sip-ip and ext-rtp-ip and deal with setting local-network-acl to the proper network/mask because you're answering with RFC1918 media IP's to the cisco… its retarded and will just start sending to those IP's and they'll usually never make it back to you. >> >> /b >> >> On Jul 19, 2011, at 7:16 PM, Mitch Johnson wrote: >> >>> I have a connection between my FreeSWITCH and a Cisco CallManager: >>> >>> This is from the default.xml >>> >>> <extension name="Dial to CM8"> >>> <condition field="destination_number" expression="^(800\d)$"> >>> <action application="bridge" data="sofia/internal/$1(a)172.16.200.100"/> >>> </condition> >>> </extension> >> >

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Altering arbitrary channels
by Isaac Jurado 25 Jul '11

25 Jul '11

Hi all, >From the diaplan, is it possible to set a variable in any other channel? That is, by specifying the channel UUID. Thanks. -- Isaac Jurado Internet Busines Solutions eConcept

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memory leak
by Tihomir Culjaga 25 Jul '11

25 Jul '11

hello, i upgraded to the latest GIT compiled configured FS without any issues... everything looks great... but when i put this into some traffic i noticed FS uses more and more memory... until it finished into swap. I didn't have issues like that previously but unfortunately i cannot recall what was the version with no issues. The server is 64bit CentOS 5.4 and the FS version is the latest (well 4 days old tops). Im using just the standard modules (the list provided below) and FS keeps leaking memory <load module="mod_console"/> <load module="mod_logfile"/> <load module="mod_enum"/> <load module="mod_cdr_csv"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_loopback"/> <load module="mod_commands"/> <load module="mod_conference"/> <load module="mod_db"/> <load module="mod_dptools"/> <load module="mod_expr"/> <load module="mod_fifo"/> <load module="mod_hash"/> <load module="mod_voicemail"/> <load module="mod_esf"/> <load module="mod_fsv"/> <load module="mod_cluechoo"/> <load module="mod_valet_parking"/> <load module="mod_dialplan_xml"/> <load module="mod_dialplan_asterisk"/> <load module="mod_spandsp"/> <load module="mod_g723_1"/> <load module="mod_g729"/> <load module="mod_amr"/> <load module="mod_ilbc"/> <load module="mod_speex"/> <load module="mod_h26x"/> <load module="mod_siren"/> <load module="mod_sndfile"/> <load module="mod_native_file"/> <load module="mod_local_stream"/> <load module="mod_tone_stream"/> <load module="mod_spidermonkey"/> <load module="mod_lua"/> <load module="mod_say_en"/> This is the dialplan im triggering... <extension name="doRedirect"> <condition field="destination_number" expression="^300030.*$"> <action application="info"/> <action application="redirect" data="<sip:0999901234567889@10.1.1.11:5060;user=phone>;q=0.99"/> <action application="hangup"/>  </condition> </extension> I have i have tried a pure "hello world" dialplan as: <extension name="testDP"> <condition field="destination_number" expression="^300030.*$"> <action application="info"/> <action application="hangup"/> </condition> </extension> But the situation is exactly the same. This is the behavior i'm having .... memory just keep growing and growing until it hits the swap where the server is doomed. tail -f smaps_log.log (this is from cat /proc/$PID/smaps | grep heap) TIM,SIZE,RSS,SHARED_CLEAN,SHARED_DIRTY,PRIVATE_CLEAN,PRIVATE_DIRTY,SWAP 2011-07-13 13:50:34,19268,18916,0,0,0,18916,0 2011-07-13 13:51:34,19268,18972,0,0,0,18972,0 2011-07-13 13:52:34,20324,20088,0,0,0,20088,0 2011-07-13 13:53:34,21264,20920,0,0,0,20920,0 2011-07-13 13:54:34,21048,20856,0,0,0,20856,0 2011-07-13 13:55:35,22396,22076,0,0,0,22076,0 2011-07-13 13:56:35,22396,22164,0,0,0,22256,0 2011-07-13 13:57:35,23220,22936,0,0,0,22936,0 2011-07-13 13:58:35,23480,23288,0,0,0,23288,0 $ tail -f mem.log (this is from ps) TIMESTAMP PID RSS %MEM 2011-07-13 13:48:55 2405 65144 12.8 2011-07-13 13:49:55 2405 66932 13.2 2011-07-13 13:50:55 2405 69576 13.7 2011-07-13 13:51:55 2405 71552 14.1 2011-07-13 13:52:55 2405 73576 14.5 2011-07-13 13:53:55 2405 75804 14.9 2011-07-13 13:54:55 2405 78388 15.4 2011-07-13 13:55:55 2405 80544 15.9 2011-07-13 13:56:55 2405 81784 16.1 2011-07-13 13:57:55 2405 83720 16.5 2011-07-13 13:58:55 2405 86524 17.0 FS runs under root account and it is started as: /usr/local/freeswitch/bin/freeswitch -nc -nonat -db /dev/shm -conf /usr/local/freeswitch/conf/ -log /usr/local/freeswitch/log/ -run /usr/local/freeswitch/run/ Also ulimit is like this; ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 240 ulimit -l unlimited so, does anyone have a clue on why the memory goes up ? ... what I'm doing wrong ? Regards, Tihomir.

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Re: [Freeswitch-users] One way calling issue over an internal trunk
by Mitch Johnson 25 Jul '11

25 Jul '11

Sorry for the delayed response, got tied up in other things. A quick recap. I can dial from my call manager to Freeswitch but cannot dial from freeswitch to call manager. I wasn't sure what you meant by setting the ext-sip-ip and ext-rtp-ip. So I replaced the auto-nat with local-network-acl. I'll always be on the internal network so should never need the nat. I saw in the acl.conf.xml that it automatically populates all my local subnets. The networks are correct under the domains of the acl.conf. I'm not sure where else to look. Any help would be greatly appreciated. Thanks so much, Mitch On Jul 19, 2011, at 11:29 PM, freeswitch-users-request(a)lists.freeswitch.org wrote: > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] One way calling issue over an internal trunk > Date: July 19, 2011 8:55:02 PM EDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > you need to set the ext-sip-ip and ext-rtp-ip and deal with setting local-network-acl to the proper network/mask because you're answering with RFC1918 media IP's to the cisco… its retarded and will just start sending to those IP's and they'll usually never make it back to you. > > /b > > On Jul 19, 2011, at 7:16 PM, Mitch Johnson wrote: > >> I have a connection between my FreeSWITCH and a Cisco CallManager: >> >> This is from the default.xml >> >> <extension name="Dial to CM8"> >> <condition field="destination_number" expression="^(800\d)$"> >> <action application="bridge" data="sofia/internal/$1(a)172.16.200.100"/> >> </condition> >> </extension> >

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Sofia stack sip rfc conformance.
by Narendra Sirugudi 24 Jul '11

24 Jul '11

Hi All, I wanted to know what all SIP standards/rfc(s) does Sofia stack conform to ? I could not find this informantion anywhere so far. thanks, --naren

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Using chat command in a dialplan
by Kurtis Heimerl 23 Jul '11

23 Jul '11

I feel somewhat embarrassed, but I can't figure out how to cause a chat event to go off from the dialplan. I've tried: <action application="chat" data="sip|${username}|${destination_number}@${domain_name}|${msg_body}"/> and <action application="eval" data="chat sip|${username}|${destination_number}@${domain_name}|${msg_body}"/> and neither worked. The wiki says it's part of the API (http://wiki.freeswitch.org/wiki/Mod_dptools) but I'm pretty sure that's just a command-line API. Is there a way to run a command-line call from the dialplan? Any direction would be appreciated.

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Call disconnect after transfer
by Mike Burlingame 23 Jul '11

23 Jul '11

I am having a small issue with calls getting disconnected after transfer from a customer signaling is posted at http://pastebin.freeswitch.org/16869 but basic call flow is call is sent to user - user answers - user transfer the call to another user - invite is sent to Freeswitch with out SDP - 200 OK provided to customer with SDP - ACK received from customer with SDP - Freeswitch disconnects call with [INCOMPATIBLE_DESTINATION] I have 3PCC enabled in FreeSwitch via sofia.conf.xml <param name="enable-3pcc" value="true"/>

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ftmod_libPRI outgoing calls channel selection
by Michal Zubáč 23 Jul '11

23 Jul '11

Hi. I've noticed that outgoing call channels in freetdm/ftmod_libpri are not selected exclusively in Q.931 SETUP messages. Channel identification "exclusive" bit is 0 so it only means "preferred". In some cases freeswitch selected channel which was not ready for dialing yet and ISDN operator answered with another channel in PROGRESS message. So actual call state messages were passed trough another channel and it resulted with lots of VETO and consecutive RESET messages, because of channel state confusion in FreeSwitch. I think that it's better to force channel selection and in some cases choose wrong channel and get info about it. It's definitely better, then choose something, assume that it is correct although actual communication is done on another channel and such calls are "doomed"... I've attached patch, in which we force selection of ISDN channel. Maybe better solution will be some configuration variable, which will control this behaviour. Consider. Does anybody have similar experience and if so, how do you deal with it? Or am I mistaken in some of my conclusions? -- Michal Zubac ComGate Interactive s.r.o. Prague Marina Office Center Jankovcova 1596/14a 17000 Praha 7, Czech Republic

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Can't get fax working
by Boris Kovalenko 23 Jul '11

23 Jul '11

Hello! I need help with fax configuration. My network is: PSTN E1 --- Cisco AS5350 -- Freeswitch -- Audiocodes MP-114 -- Fax FS version: FreeSWITCH Version 1.0.head (git-062f07f 2011-07-16 12-35-25 +0200) Audiocodes firmware: 6.00A.038.004 I can't get faxes working :( In FS profiles there is a parameter <param name="t38-passthru" value="true"/> Cisco DP fax settings: fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711alaw no vad On Audiocodes there is also T38 enabled (as I think) but there is no T38. There are messages on MP-114: ErrMgs=16 T38Decoder received Non-T38 Packet and || Modem Relay Is Not Supported! Forcing Bypass Mode Please, help! What information I need to collect to help solve the problem? -- Regards, Boris

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