August 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

What is the difference between preAnswer and ring_ready?
by Max Bridgewater 31 Aug '09

31 Aug '09

Hi, Assuming an inbound call, I have trouble understanding what the supposed difference between the following two set of instructions is: session.execute("ring_ready"); session.set("set", "ringback=/home/ring.wav"); and session.preAnswer(); session.streamFile("/home/ring.wav") In practice, however, the first scenario doesn't work for me although the example here suggests otherwise: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready Any idea? Thanks, Max.

4 3

How can it be done with freeswitch?
by Nagalenoj 31 Aug '09

31 Aug '09

Dear friends, I am using freeswitch for the last one month and trying to write perl scripts to do some tasks. I am trying a new thing, which is something that I don't how to do it and I don't whether it can be done with freeswitch. my requirement is as follows, A calls B[This extension has to execute a script]. The script has to do the following, * Cut the call and callback to the number. * Get a phone number[DTMF] from A. * Dial the number(C) and bridge A and C. * When C cuts the call, the control has to comeback to script. * Ask another number to A to dial out, get another phone number[DTMF] from A. * Dial the number(D) and bridge A and D. Please, help me to do this. Is there any other documentation to read about the perl freeswitch functions. -- View this message in context: http://www.nabble.com/How-can-it-be-done-with-freeswitch--tp25184845p251848… Sent from the Freeswitch-users mailing list archive at Nabble.com.

2 3

liverpie as gui-proxy?
by tom 31 Aug '09

31 Aug '09

hi, i was actually looking for something completely else, but reading wiki led me to the tought: if mod_event can basically do all kinds of FS-things, why not having a standalone ruby-on-rails as as gui which interacts with the evnet-socket via liverpie? im a total newbie to FS, just writing down my thoughts... thx for ur replies

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external Gateways
by NOx-WHV 31 Aug '09

31 Aug '09

Hello, i use a sip gateway witch is only be able to allocate one sip channel. How can I bridge a call to a second gateway if the first gateway is in use. Now the dialplan look like this: <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^04(\d+)$"> <action application="set" data="effective_caller_id_number=300"/> <action application="set" data="sip_secure_media=false"/> <action application="bridge" data="sofia/gateway/mygateway/04$1(a)XXX.XXX.XXX.XXX"/> </condition> Thanks for help -- View this message in context: http://www.nabble.com/external-Gateways-tp25223286p25223286.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
by Hristo Benev 31 Aug '09

31 Aug '09

Now it is correct. Thank you for your time. >-------- Оригинално писмо -------- >От: rod >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >До: freeswitch-users(a)lists.freeswitch.org >Изпратено на: Събота, 2009, Август 29 08:43:24 EEST >Ok, > >I found what's happening. I probably did some change on the wiki to >start reflect the new configuration, without having time enough to check >the configuration. There is a mistake in your dialplan configuration. >You should put this instead: > >wrong line: > > > >please correct with: > > > >This should match PEER_01 in dialplan instead of trying matching >France@PEER_01. > >Let me know if this is right now. > >rod > >Hristo Benev a écrit : >> Hello Rod, >> >> I did the change. >> >> Here is extract of console: >> ----------------------------------------------------------------------------------------- >> variable_continue_on_fail: [true] >> variable_sip_h_X-ROUTE: [LOOKUP] >> variable_export_vars: [sip_h_X-ROUTE] >> variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20] >> variable_sip_redirect_contact_0: [sip:France@PEER_01] >> variable_sip_redirected_to: [sip:France@PEER_01] >> variable_sip_redirect_contact_user_0: [France] variable_sip_redirect_contact_host_0: [PEER_01] >> variable_sip_redirect_dialstring_0: [sofia/internal/sip:France@PEER_01] >> variable_sip_redirect_dialstring: [sofia/internal/sip:France@PEER_01] >> variable_proto_specific_hangup_cause: [sip:503] >> variable_sip_hangup_phrase: [DNS Error] >> variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE] >> variable_ROUTE_GW: [France] >> variable_AREA: [France] >> variable_current_application: [info] >> ------------------------------------------------------------------------------------------ >> I have different value it is actually the description field as shown here: >> >> -------------------------------- >> /opt/kamailio/sbin/kamctl cr show >> cr carrier names >> +----+---------+ >> | id | carrier | >> +----+---------+ >> | 1 | default | >> +----+---------+ >> cr domain names >> +----+---------+ >> | id | domain | >> +----+---------+ >> | 1 | default | >> +----+---------+ >> cr routes >> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ >> | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description | >> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ >> | 1 | 1 | 1 | 1000 | 0 | 0 | 1 | 0 | PEER_01 | | | France | >> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ >> ----------------------------------------- >> >> And here is what I have in kamailio: >> ------------------------------------------------------------------------------------- >> ####### Routing Logic ######## >> >> >> # main request routing logic >> >> route{ >> >> if (!mf_process_maxfwd_header("10")) { >> sl_send_reply("483","Too Many Hops"); >> exit; >> } >> >> t_check_trans(); >> >> if ($rU==NULL) { >> # request with no Username in RURI >> sl_send_reply("484","Address Incomplete"); >> exit; >> } >> >> # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP >> if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){ >> if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){ >> #xlog("ROUTING FAILED: no route found for $rU"); >> sl_send_reply("604", "Unable to route this call"); >> exit; >> } else { >> xlog("LOOKUP FOUND: $rd $avp(s:route_desc)"); >> avp_pushto("$ru/username", "$avp(s:route_desc)"); >> sl_send_reply("302", "$rd"); >> exit; >> } >> } >> } >> ---------------------------------------------------------------------------------- >> >> Another question... >> In that part of FreeSwitch dialplan.xml >> >> ----------------------------------- >> >> >> >> >> >> >> ----------------------------------- >> >> X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true. >> >> >> As for thanks - for sure by default they are also for the developers of both apps. >> I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with. >> >> >> >> >-------- Оригинално писмо -------- >> >От: rod >> >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >> >До: freeswitch-users(a)lists.freeswitch.org >> >Изпратено на: Петък, 2009, Август 28 09:54:00 EEST >> >> >Hello, >> > >> >the trace seems good. >> >If you check the answer from Kamailio, you'll see that Kamailio answers >> >with "302 PEER_01". >> > >> >As Michael Collins stated before, you can get the variable containing >> >"PEER_01", then this variable is stored in a custom variable. >> > >> >In your dialplan, may you please add: >> > >> >, just before the transfer line, eg: >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> >Using application Info, you'll see on the console (or CDR) the list of >> >variables used for this call. You should see the content of >> >"${sip_redirect_contact_user_0}" that should contain the value >> >"PEER_01". Please check this and let me know. >> > >> >For the new configuration file, no problem for sharing. But as I wrote >> >on the wiki page, I worked on this setup cause LCR module was not >> >available when I start working on FS, nor I'm a good programmer to write >> >a server side HTTP script (used by xml_curl) that could scale to my needs. >> > >> >I enhanced a bit this configuration with support for fallback routing, >> >almost realtime graph (every minutes using www.cacti.net and some >> >functions in FS like limit_hash) of number of concurrent calls per AREA, >> >PEER... >> > >> >Thanks for "the good tutorial", but don't forget the dev team who did >> >this great product ;-) >> > >> >rod. >> > >> > >> >Hristo Benev a écrit : >> >> I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080). >> >> >> >> If you need additional info I'll provide it. >> >> >> >> Here is trace: >> >> >> >> ngrep -d any -nn -i '1000' port 5062 -W byline >> >> interface: any >> >> filter: (ip or ip6) and ( port 5062 ) >> >> match: 1000 >> >> # >> >> U 10.10.10.10:5090 -> 127.0.0.1:5062 >> >> INVITE sip:1000@127.0.0.1:5062 SIP/2.0. >> >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj. >> >> Max-Forwards: 69. >> >> From: "Extension 1001" ;tag=978g69jZaFpBD. >> >> To: . >> >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. >> >> CSeq: 119574771 INVITE. >> >> Contact: . >> >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported. >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. >> >> Supported: timer, precondition, path, replaces. >> >> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. >> >> Content-Type: application/sdp. >> >> Content-Disposition: session. >> >> Content-Length: 429. >> >> X-ROUTE: LOOKUP. >> >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off. >> >> . >> >> v=0. >> >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10. >> >> s=FreeSWITCH. >> >> c=IN IP4 10.10.10.10. >> >> t=0 0. >> >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13. >> >> a=rtpmap:0 PCMU/8000. >> >> a=rtpmap:115 G7221/32000. >> >> a=fmtp:115 bitrate=48000. >> >> a=rtpmap:107 G7221/16000. >> >> a=fmtp:107 bitrate=32000. >> >> a=rtpmap:9 G722/8000. >> >> a=rtpmap:8 PCMA/8000. >> >> a=rtpmap:3 GSM/8000. >> >> a=rtpmap:101 telephone-event/8000. >> >> a=fmtp:101 0-16. >> >> a=rtpmap:13 CN/8000. >> >> a=ptime:20. >> >> >> >> # >> >> U 127.0.0.1:5062 -> 10.10.10.10:5090 >> >> SIP/2.0 302 PEER_01. >> >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj. >> >> From: "Extension 1001" ;tag=978g69jZaFpBD. >> >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8. >> >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. >> >> CSeq: 119574771 INVITE. >> >> Contact: sip:France@PEER_01. >> >> Server: Kamailio (1.5.2-notls (i386/linux)). >> >> Content-Length: 0. >> >> . >> >> >> >> # >> >> U 10.10.10.10:5090 -> 127.0.0.1:5062 >> >> ACK sip:1000@127.0.0.1:5062 SIP/2.0. >> >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj. >> >> Max-Forwards: 69. >> >> From: "Extension 1001" ;tag=978g69jZaFpBD. >> >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8. >> >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. >> >> CSeq: 119574771 ACK. >> >> Content-Length: 0. >> >> . >> >> >> >> >> >> >> >> >> >> >> >> >-------- Оригинално писмо -------- >> >> >От: Hristo Benev >> >> >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >> >> >До: freeswitch-users(a)lists.freeswitch.org >> >> >Изпратено на: Четвъртък, 2009, Август 27 14:58:53 EEST >> >> >> >> > >> >> >Bojnour, >> >> > >> >> >I'll send a trace ASAP. >> >> > >> >> >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config) >> >> > >> >> >I'm really interested to see the updates of the project. >> >> > >> >> >Thank you for the good tutorial. >> >> > >> >> >Hristo >> >> > >> >> > >> >> > >-------- Оригинално писмо -------- >> >> > >От: rod >> >> > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >> >> > >До: freeswitch-users(a)lists.freeswitch.org >> >> > >Изпратено на: Четвъртък, 2009, Август 27 09:41:05 EEST >> >> > >> >> > >Hi Hristo, >> >> > > >> >> > >I'm the author of this setup and wiki page. I did a lot of modifications >> >> > >on this setup (alternative routing if failure essentially) but don't >> >> > >have too much time to update the wiki. >> >> > > >> >> > >May you please send me an ngrep trace when you call 1000: >> >> > > >> >> > >ngrep -d any -nn -i '1000' port 5060 -W byline >> >> > > >> >> > >I will check what's happening. >> >> > >Do you have an entry for 1000 in your mysql database ? >> >> > > >> >> > >regards, >> >> > >rod >> >> > > >> >> > >Hristo Benev a écrit : >> >> > >> >> >> > >> It seems that the problem is on kamailio configuration. >> >> > >> Will ask on their list. >> >> > >> >> >> > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required. >> >> > >> >> >> > >> I have it setup as friend in asterisk, but still ??? >> >> > >> >> >> > >> Any ideas? >> >> > >> >> >> > >> Thanks, >> >> > >> >> >> > >> >> >> > >> >-------- Оригинално писмо -------- >> >> > >> >От: Hristo Benev >> >> > >> >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >> >> > >> >До: freeswitch-users(a)lists.freeswitch.org >> >> > >> >Изпратено на: Сряда, 2009, Август 26 22:02:13 EEST >> >> > >> >> >> > >> > I think that the problem is here: >> >> > >> >------------------------- >> >> > >> >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING >> >> > >> >Dialplan: sofia/internal/1001(a)209.71.254.33 parsing [ROUTING->PEER_01] continue=false >> >> > >> >Dialplan: sofia/internal/1001(a)209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false >> >> > >> >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting >> >> > >> >-------------------------- >> >> > >> > >> >> > >> >Actually Regex FAIL >> >> > >> > >> >> > >> >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success? >> >> > >> >Here is my default.xml: >> >> > >> >---------------- >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> >-------------------------- >> >> > >> > >> >> > >> > >> >> > >> > >> >> > >> > >-------- Оригинално писмо -------- >> >> > >> > >От: Brian West >> >> > >> > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >> >> > >> > >До: freeswitch-users(a)lists.freeswitch.org >> >> > >> > >Изпратено на: Сряда, 2009, Август 26 19:47:37 EEST >> >> > >> > >> >> > >> > >We do not blindly follow 302's as that is a dangerous thing to do. >> >> > >> >You have to process all 302's in the dialplan. >> >> > >> >Set this on your sofia profile >> >> > >> >You can set these variables sip_redirect_profile, >> >> > >> >sip_redirect_context, >> >> > >> >sip_redirect_dialplan, >> >> > >> >When a redirect happens you get these variables - sip_redirect_contact_%d, >> >> > >> >sip_redirected_to, >> >> > >> >sip_redirect_contact_user_%d, >> >> > >> >sip_redirect_contact_host_%d, >> >> > >> >sip_redirect_contact_params_%d, >> >> > >> >sip_redirect_dialstring_%d, >> >> > >> >sip_redirect_dialstring, >> >> > >> >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the >> >> > >> >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems? >> >> > >> > > >> >> > >> > >> >> > >> >_______________________________________________ >> >> > >> >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 >> >> > >> > >> >> > >> >> >> > >> _______________________________________________ >> >> > >> 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 >> >> > >> >> >> > > >> >> > >_______________________________________________ >> >> > >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 >> >> > > >> >> > >> >> >_______________________________________________ >> >> >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 >> >> > >> >> >> >> _______________________________________________ >> >> 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 >> >> >> > >> >_______________________________________________ >> >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 >> > >> >> _______________________________________________ >> 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 >> > >_______________________________________________ >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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Question about mod_limit
by Anatoliy Kounitskiy 31 Aug '09

31 Aug '09

I have question about mod_limit: is there a reason to have difference between the api applications and the dialplan? I mean when you use db function from the cli you can use insert/select/delete but when you use it from the dialplan you can only use insert/delete. #define DB_USAGE "[insert|delete]/<realm>/<key>/<val>" #define DB_DESC "save data" SWITCH_STANDARD_APP(db_function) but in the SWITCH_STANDARD_API(db_api_function) i have three options insert,delete and select What i try to accomplish is to be able to insert values from the dialplan and after that to be able to use them ( to select and use the result) from another extension. Thank you in advanced, -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy(a)kounitskiy.com Mobile: +359898913540

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Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
by Pete Mueller 31 Aug '09

31 Aug '09

2 1

Transferring to another extension is VERY slow.
by Erik Wickstrom 30 Aug '09

30 Aug '09

Hi, I'm having some trouble debugging why the transfer command is so slow in my case. I have a lua IVR setup which responds to user selections by transferring the call to another dialplan ext. Pretty basic stuff. The problem is, it takes 30-40 seconds for the the transfer to actually start after the key press. After about 30 seconds of silence, "hold music" finally starts and the other extension finally answers shortly thereafter. I've tried transferring to endpoint extensions, the 5000 sample IVR, 9999 hold music -- in all cases there is this long pause. Does anyone know what could be causing this? Thanks! Erik

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Troubles build with mod_cepstral
by Max Bridgewater 30 Aug '09

30 Aug '09

Hi, I was trying to build freeswitch with cepstral enabled. Here is the procdure i am following: http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_v… Unfortunately, the build fails as show in the log message hereunder. In a previous thread, Brian says that the SDK is needed. Yet on the above wiki page, it is said that the SDK is not needed. Any suggestion? Max. Compiling mod_cepstral.c ... Creating mod_cepstral.so... /usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when searching for -lswift /usr/bin/ld: cannot find -lswift collect2: ld returned 1 exit status gcc -w -I/home/installs/freeswitch/src/include -I/home/installs/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -I/include -I/opt/swift/include -shared -o .libs/mod_cepstral.so -shared -Wl,-x .libs/mod_cepstral.o -L/lib -L/opt/swift/lib -lswift -lm /home/installs/freeswitch/.libs/libfreeswitch.so -L/home/installs/freeswitch/libs/apr-util/xml/expat/lib /home/installs/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /home/installs/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt -lpthread -L/home/installs/freeswitch/libs/srtp -L/usr/kerberos/lib64 -lssl -lcrypto -ldl -lz -lncurses -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod make[5]: *** [mod_cepstral.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cepstral-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2

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delay buildup in conference
by Public Dump 30 Aug '09

30 Aug '09

When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ? I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module. Any advice? Thanks

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