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mod_python docs & Wednesday BSD maintainer call
by Brian West 11 Feb '14

11 Feb '14

Dear FreeSWICHers, Anyone using mod_python? Want to assist in updating the mod_python documentation on the wiki? In addition this Wednesday call is about *BSD maintenance. Its at the usual sip:888@conference.freeswitch.org, if you have questions prior to the meeting you can either ask in response to this email or contact me off list… or at the numbers below. Hope to hear everyone at the Wednesday call…. Also if the time isn’t good for you we can have a second call at a more reasonable hour if you happen to be on the other side of this spinning ball of dirt. ;) -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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no lock file
by Joli Martinez 10 Feb '14

10 Feb '14

Hello, I am trying to create an init script for CentOS 6.5. Freeswitch runs fine from "/usr/local/freeswitch/bin/freeswitch". When I attempt to fix the redhat script to my install it doesnt start. I found the problem to be the lock file. My freeswitch install doesnt seem to put the lock file in "/var/lock/subsys/freeswitch". I cant seem to find where it puts it by default. I have done a grep and cant find it anywhere. I installed freeswitch by compliling it I am running 1.2.18+git~20140122T234636Z~454a288ec6~64bit (git 454a288 2014-01-22 23:46:36Z 64bit. I have not changed any of the default setting when I installed FS. Where is the lock file located? Thanks

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Can FS benefit from multicores processor Linux?
by Donny Hardyanto 10 Feb '14

10 Feb '14

Hi, I need help in best practice using FS. My question is single installation box of FS can gain benefit from the multicore processor and huge memory? Like the performance and capacity of FS will increase from 1 core to 4 cores or even 16 cores? Or if not, can we still gain benefit by using multple VM (each 1 core) in one box? We are talking in Linux of course, using Ubuntu or CentOS. Of cource I am not asking for maximum capacity, because it will be vary between implementation and I know we all hate that question (because what follow after that :) ). Thanks, Donny

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Way to figure out all possible prams/values for mod_sofia for example
by Muhammad Naseer Bhatti 10 Feb '14

10 Feb '14

Hi, is there a way to figure out all possible params and their possible values and their types for any given mod? We are working on an interface which will control all possible config params for FreeSWITCH. We are currently going through the source to know what possible values for each param. For example, caller-id-type have possible values of rpid, pid and none. dtmf-type may have rfc2833, info or none. These two prams are fixed params, and anything else as their value than the one mentioned will not work. This is just an example. Thanks, Muhammad

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G729.A on ARM
by Lesley Pervis 10 Feb '14

10 Feb '14

Does the project have any plans to sell G729 codecs for AMR as well as for x86_64/i386? My company has been running a FS VoIP PA endpoint on ARM, kind of like Ken's Switch Pi, but simpler, and we'd really like to put G729 on the things, if possible.

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esl, send_dtmf, premature ESL_EVENT_CHANNEL_EXECUTE_COMPLETE
by Marcin S 10 Feb '14

10 Feb '14

Hello, In my simple C application I want to send dtmfs. This is quite easy - you just call esl_execute("send_dtmf", "1234", uuid); and it works. However, ESL returns ESL_EVENT_CHANNEL_EXECUTE_COMPLETE way to fast - even before it starts dtmfs generation! How can I know, when dtmfs were really generated? So far, the only method is to issue some sleep(), but it looks like big workaround.

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Mod_Manage won't build switch_core_port_allocator_new
by Andre 10 Feb '14

10 Feb '14

Hi, I downloaded freeswitch-1.2.19.tar.gz today and Mod_manage won't build. The code that is the problem is line 8794 in freeswitch_wrap.2010.cxx SWIGEXPORT int SWIGSTDCALL CSharp_switch_core_port_allocator_new(unsigned short jarg1, unsigned short jarg2, unsigned long jarg3, void * jarg4) { int jresult ; switch_port_t arg1 ; switch_port_t arg2 ; switch_port_flag_t arg3 ; switch_core_port_allocator_t **arg4 = (switch_core_port_allocator_t **) 0 ; switch_status_t result; arg1 = (switch_port_t)jarg1; arg2 = (switch_port_t)jarg2; arg3 = (switch_port_flag_t)jarg3; arg4 = (switch_core_port_allocator_t **)jarg4; result = (switch_status_t)switch_core_port_allocator_new(arg1,arg2,arg3,arg4); jresult = result; return jresult; } error C2664: 'switch_core_port_allocator_new' : cannot convert parameter 1 from 'switch_port_t' to 'const char *' ..\mod_managed\freeswitch_wrap.2010.cxx IntelliSense: too few arguments in function call ..\mod_managed\freeswitch_wrap.2010.cxx 8806 79 mod_managed Looking at the method switch_core_port_allocator_new this seems to of changed. How do I fix the code? SWITCH_DECLARE(switch_status_t) switch_core_port_allocator_new(_In_ const char *ip, _In_ switch_port_t start, _In_ switch_port_t end, _In_ switch_port_flag_t flags, _Out_ switch_core_port_allocator_t **new_allocator);

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Performance gain for Asterisk 12 to use pjsip at core?
by Henry Huang 09 Feb '14

09 Feb '14

Asterisk 12 has added a new sip driver using pjsip. I remember reading about the discussion why FreeSWITCH chose to use sofia instead of other sip stacks, but I can't find it anymore. So my question is, do you guys think Asterisk will be able to boost their performance by swapping out their sip core to use pjsip and be somewhat on par with FreeSWITCH performance wise? Thanks, Henry

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FreeSwitch Proxy + RTPProxy Media server
by adahary 09 Feb '14

09 Feb '14

I would like to integrate FS with RTPproxy like openSips and kamailio are well integrated with it. FS should handle the SIP signaling and the RTPproxy should relay the RTP stream from A to B: A.sip <=> FS <=> B.sip FS = PASS-THRU A.rtp <=> RTPproxy <=> B.rtp. I understand that FS should ask the RTPproxy to allocate UDP ports for both endpoint and then pass-thru-bridge them to cummunicate directly through the RTPproxy. What I cannot yet figure out is how replace both A and B ip/port sets in SDP with the RTPproxy ip/port-udp. I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't figure it out. how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when A calls B? Tried the following with no help. <action application="set"> </action> <action application="set"> </action> Any tip/help/advise will be appriciated. Regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Med… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch not putting external IP in SD and From Header when dialing out SIPURI by external profile
by Andrew Paul 09 Feb '14

09 Feb '14

Hi all, ------------------FREESWITCH--------------(bridge=sofia/external/sip:outbound-opensip.com:5060 )-----------OPENSIP---------------------- I am facing NAT problem when i am dialing SIP URI via sip profile. When i am dialing the SIP uri with Gateway trunk NAT is working properly. Dialing SIP URI <action application="bridge" data="sofia/external/sip: outbound-opensip.com:5060" In external profile setting I made the setting for ext-rtp-ip and ext-sip-ip. In freeswitch cli "sofia status profile external" i am able to see the external rtp and sip ip correctly. But when freeswitch sending INVITE external IPs are not set in SDP and FROMHeader, instead it is using local IP. When i tried to debug the problem in switch_core_media.c freeswitch failed to check the NAT settings. switch_core_media.c SWITCH_DECLARE(switch_status_t) switch_core_media_choose_port(switch_core_session_t *session, switch_media_type_t type, int force) /* Check if NAT is detected */ if (!zstr(smh->mparams->remote_ip) && switch_core_media_check_nat(smh, smh->mparams->remote_ip)) { /* Yes, map the port through switch_nat */ /* Code */ } else { /* No NAT traversal required, use the profile's rtp ip */ use_ip = smh->mparams->rtpip; } Here remote IP is NULL always when i am dialing URI with external profile. When I tried to make call by "sofia/gateway/trunk/sip: outbound-opensip.com:5060" i can see the remote ip as "oubound-opensip.com" and its able to external IP in SDP. Can any body tell how to solve this problem ? Is there any configuration to set relam in dialplan ? I am using freeswitch version 1.5.6. Hopping the earliest response. Regards Andrew

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