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DNS SRV in bridge, problems
by Miha 11 Dec '13

11 Dec '13

Hi, I have in a bridge dns srv domain which works right. When server with lower priorty is up, FS sends invite's to it which is ok. But when primary/lower priority server is down FS still send's invite's to it. All UAC re-register to server with higher priory which is only avalible. Are there any other configuration to be made on fs, so that FS will sends invites to right server? tnx miha

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Recording G729
by Kees Varekamp 11 Dec '13

11 Dec '13

Hi all, I'm suffering from poor sound quality on my setup, using GSM on the a-leg and PCMU on the b-leg. I'm thinking it's mostly bandwidth/network related, so I'm considering switching to G729 on both legs, as that is the only compressed codec that the b-leg supports (it's a sip gateway). But a complicating factor is that the calls need to be recorded in mp3. So I have some questions that I hope someone here can answer: - Does transcoding (for example GSM<>PCMU) negatively affect the sound quality, even when the CPU seems to be ok with it? - If yes, does recording 2 legs using the same codec (say G729) to a different sound format (say mp3) have the same (negative) effect? - When you use G729 on both legs (using mod_com_g729), it is even possible to record to mp3? Thank you, Kees

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Luasocket
by Neil Patel 11 Dec '13

11 Dec '13

I have built and installed Luasocket 2.0.2 as per the instructions on the wiki <http://wiki.freeswitch.org/wiki/Installing_LuaSocket>. I am running FS 1.2 stable. But as has been <http://jira.freeswitch.org/browse/FS-4712> noted<http://jira.freeswitch.org/browse/FS-2893>in Jira, Luasocket causes FS to crash from time to time. In particular socket.http.request() and post(). Has anyone been able to use Luasocket in a stable environment without FS crashing? What luasocket version? Any special build instructions? I would be happy to update the wiki with working instructions. Thanks, Neil

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Possible bug in playback: local_stream://moh
by Nuno Reis 11 Dec '13

11 Dec '13

Hi guys. I've just noticed that playback of local_stream://moh stopped working after upgrading from FS 1.2.3 to 1.2.14 Here's the tail output from the dailplan in FS 1.2.3 (everything works): ... EXECUTE sofia/internal/nmreis@domain1_2_3 playback(local_stream://moh) 2013-12-06 00:54:42.099381 [DEBUG] mod_local_stream.c:417 Opening Stream [moh/8000] 8000hz 2013-12-06 00:54:42.099381 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 20ms ... Here's the tail output from the dailplan in FS 1.2.14 (everything still seems to be ok but i can't hear any sound): ... EXECUTE sofia/internal/nmreis@domain_1_2_14 playback(local_stream://moh) 2013-12-06 00:57:46.057502 [DEBUG] mod_local_stream.c:498 Opening Stream [moh/8000] 8000hz 2013-12-06 00:57:46.057502 [DEBUG] switch_ivr_play_say.c:1319 Codec Activated L16@8000hz 1 channels 20ms ... This issue is happening accross several FSs running 1.2.14. The only FSs where i'm able to hear moh are on the old 1.2.3 versions. Anyone experiencing the same thing? Any hints? Looking forward to hear from you. BR, -- *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/>** <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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Help Invite User Directory
by FERNANDO VILLARROEL 11 Dec '13

11 Dec '13

Dear All. I am confused with my user directory config I do not understand the follow: I am trying to config FS on Public IP addresse like 190.xxx.xx.xxx and fs and log show me: 2013-12-11 00:41:39.095757 [DEBUG] sofia.c:8066 IP 200.104.24.129 Rejected by acl "domains". Falling back to Digest auth. 2013-12-11 00:41:39.095757 [WARNING] sofia_reg.c:1634 SIP auth challenge (INVITE) on sofia profile 'internal' for [09999(a)190.xxx.xx.xxx] from ip 200.xxx.xx.xxx and i see that the calls are processed by public context 2013-12-11 00:41:39.155760 [INFO] mod_dialplan_xml.c:558 Processing 5644 <5644>->09999 in context public Why the calls are not processed by default context like say user config directory? My vars.xml <X-PRE-PROCESS cmd="set" data="domain=190.xxx.xx.xxx"/> <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/> My ~/conf/directory/default.xml <domain name="$${domain}"> <params> And when i try to set some variables like ${accountcode} this var did not set How i can solve my problem, i appreciated your advice Regards

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variable_originate_signal_bond vs variable_signal_bond
by Antonio Silva 10 Dec '13

10 Dec '13

Hi, I've a small dialer that uses loopback and i'm trying to find out the uuid for the real channel sofia when the B channel is answered. So far i use the variable_originate_signal_bond that gives me the uuid of the sofia channel. But in some calls i don't have this variable but instead i have variable_signal_bond. In my code right now i test both to have the real channel, so there is no problem here. I notice that when i have signal_bond i don't have originate_signal_bond and is present when the bridge returns SUCCESS (originate_disposition=SUCCESS). But in some case the originate_disposition is equal failure and the call is successfully connected to sofia channel. What is the differences between this variables? or should i use another method to get the uuid for the realchannel? Thanks, António

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display name updates with two outbound channels
by Hristo Trendev 10 Dec '13

10 Dec '13

Hi, I am trying to bridge two remote endpoints via originate as described at http://wiki.freeswitch.org/wiki/Mod_commands#originate originate sofia/example/300(a)foo.com &bridge(sofia/example/400(a)bar.com) All goes well, with the exception of display name updates. The problem is that any updates from the second call leg (400(a)bar.com) are not sent to the 300(a)foo.com call leg. As soon as 400(a)bar.com sends an update I can see a message in the log that says (just an example): Update Callee ID to "Name" <123123123> However, nothing is being sent out to the first call leg (300(a)foo.com) and the endpoint doesn't update the displayed party name. Technically we have 2 *outbound* call legs in this case: leg 1: from FS to 300(a)foo.com leg 2: from FS to 400(a)foo.com so a *callee* ID update from leg 2 should translate into *caller* ID update on leg 1. I am not sure how to accomplish this. Is it possible at all? BTW, I tried setting <param name="send-display-update" value="true"/> in the sip profile, but it makes no difference. I also thought about updating the ID on call leg 1 via an api call (for example api_on_answer or similar), but I couldn't find any api/application that will allow me to actively change the ID of the call leg if it was already answered. I am trying this on v1.2.14. Regards, Hristo

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Lua script
by Joli Martinez 10 Dec '13

10 Dec '13

Hello, I am writing a Lua script that will get the callerId info. I need 1 to be able to get the callerID and also to print any messages to the FS_CLI for debugging purpose. By itself the session:execute does transfer to the correct conference, but it does not work with the callerID. And the consoleLog doesnt work it give me an error on the screen. #!/usr/bin/lua session:answer(); callerID = session:getVariable("caller_id_number"); freeswitch.consoleLog("Hello World"); --session:execute("conference","4567"); --while (session:ready() == true) do --session:setAutoHangup(false); if (callerID == 1234567890) then session:execute("conference","4567"); --else -- session:execute("transfer","7890"); end -- end

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FreeSWITCH Twitter Account @freeswitch now!
by Brian West 10 Dec '13

10 Dec '13

Dear FreeSWITCHers, Please make sure you follow @freeswitch now as we have acquired that handle on twitter. w00t! -- 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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Proxy media
by peetee23＠hushmail.com 10 Dec '13

10 Dec '13

Hi All, on wiki http://wiki.freeswitch.org/wiki/Proxy_Media is mentioned that I have to put proxy_media=true "before bridge". Could someone please explain me where exactly I have to setup this on latest stable FS with default configuration? Proxy media has to be used only during calls between users on same PBX. Thank you for you reply Kind regards, PT

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