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autofix incorrect packetization
by Hristo Trendev 17 Dec '13

17 Dec '13

Hi, I get choppy audio in conferences from endpoints that send G722@30mspackets. The endpoint in question is a FRITZ!Box CPE and this seems to be a known problem with these type of devices. They simply ignore the offered ptime in SDP and stick to sending 30ms RTP packets no matter what. The last time I tested with a FRITZ!Box CPE was in February and I am absolutely sure that FS used to auto fix this type of problems in the past and when this was happening I would see the following lines in the logs: "Asynchronous PTIME not supported, changing our end from 20 to 30" "Changing Codec from G722@20ms@8000hz to G722@30ms@8000hz" and also something to do with timer resolution being increased to deal with 30ms intervals. This doesn't happen any more (v1.2.14) and all I can hear is choppy audio from these CPEs. The only relevant lines I could find in the logs were: "mod_conference.c:3590 Setup timer soft success interval: 20 samples: 160" "switch_core_io.c:759 Engaging Read Buffer at 640 bytes vs 240" I also verified with a packet trace that FS is indeed receiving RTP packets with payload of 240 bytes, which would correspond to 30ms and is sending packets with 160 bytes of payload (20ms), so as far as I can tell this would actually classify as async ptime. I only see the problem one way - the choppy audio comes from the remote party, which is sending the 30ms packets into the conference. The remote party however still hears normal audio coming from the conference (although it is packetized and sent out at 20ms). Could it be that for some reason FS or the conference application is mistakenly treating the 30ms packets as 20ms ones when mixing the audio into the conference? I do recognize that this is caused by misbehaving end device and it is not a FS issue, but I just wanted to know if someone else is also seeing similar problems or if this can somehow be related to a recent change in the FS code? I have updated to v1.2.15 so I can retest, but I will only get a chance to do it on Friday. Best Hristo

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List Etiquette (just a reminder)
by Ken Rice 17 Dec '13

17 Dec '13

Hey guys, just a reminder (as someone just did this) Please do not hijack threads... If you have a new problem start a new thread but sending a new message to FreeSWITCH-users(a)lists.freeswitch.org If you have just subscribed to the list or have not posted to the list, chances are you are still moderated. We do this to reduce spam and it works quite well. If you post to the list and it doesn¹t come right through chances are you are moderated and one of the list mods will get to checking the mod queue shortly. Please keep in mind that most of the list mods are based in the USA. That means between 10PM and 8AM Eastern Time in the US, and on weekends there may be a delay in getting the queue processed. You don¹t have to post 4 times (and you most certainly should not post 4 times attempting to hijack other threads.) Have a Great Week! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Mod LCR SQL sub-query for better routing
by Victor Chukalovskiy 17 Dec '13

17 Dec '13

Hello, For those interested, I added a piece to mod lcr wiki. It makes sorting / routing logic better than default logic: http://wiki.freeswitch.org/wiki/Mod_lcr#Custom_SQL_with_sub-query_-_for_rea… Why it helps: Rates rates can often be given both on per-NPA or per-NPANXX level depending on the carrier and on the NPA. Moreover, some carriers may have NPANXX rate lower than the corresponding NPA rate, while others will have it inverse. Neither simple ORDER BY rate, prefix; nor ORDER BY prefix, rate; give the truly cheapest route. The LCR logic should be two-step process to accommodate this. Cheers, -Victor

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H323
by Richard Mace 17 Dec '13

17 Dec '13

HI All, Where should the file h323.conf.xml be located and how does FreeSWITCH know where's it's config files are and which ones to use? Thanks Richard

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clock sync troubles
by Javier Aristizábal 16 Dec '13

16 Dec '13

Hello folks, I would like to know if FreeTMD will allow two different clocks to be used for two different ISDN cards? The BRI running on the NTT (telco) clock and the PRI card running on its own hardware clock then pass an incoming call on the BRI card over FreeTDM to the PRI card. Would this avoid clock sync troubles? Regards -- Javier Aristizábal

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Is it possible to play a wav file to all legs at the same time?
by Ali Pey 16 Dec '13

16 Dec '13

Hello, I would like to play an announcement before call recording starts. I tried uuid_broadcast, but it plays the file on a-leg first and then after that it plays the file on the b-leg. It does not play the file on both legs at the same time. Is there a way to play a file on all legs simultaneously? Thanks, Ali Pey

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Building mod_managed - error with swig.cs
by Erik M. Devane - Comms Guy 16 Dec '13

16 Dec '13

All, It's been eighteen days since I moved my production server from Windows to CentOS, as I was recommended, and I couldn't be happier with how stable the system has been. Now I'm trying to migrate some tools that I wrote for Windows over to CentOS, and I'm having difficulty building mod_managed. What I want to do is to send an array of bytes over TCP whenever an incoming call is bridged or unbridged. Initially, I used the excellent ESL dll and the managed wrapper, and I tried using lua but support for sending byte arrays is limited. I decided that mod_managed would allow me to reuse some of my interface code, and so I followed instructions in the wiki to build mod_managed, using the same 1.2.15 tarball that I'm currently running. The error I'm getting is this: making all mod_managed swig.cs(27328,27): error CS1519: Unexpected symbol `event' in class, struct, or interface member declaration swig.cs(27328,29): error CS1519: Unexpected symbol `{' in class, struct, or interface member declaration swig.cs(27329,9): error CS1519: Unexpected symbol `{' in class, struct, or interface member declaration swig.cs(27330,55): error CS1519: Unexpected symbol `(' in class, struct, or interface member declaration swig.cs(27330,64): error CS1519: Unexpected symbol `,' in class, struct, or interface member declaration swig.cs(27330,86): error CS1519: Unexpected symbol `(' in class, struct, or interface member declaration swig.cs(27330,92): error CS1519: Unexpected symbol `)' in class, struct, or interface member declaration swig.cs(27332,4): error CS1525: Unexpected symbol `get' swig.cs(27333,6): error CS1525: Unexpected symbol `IntPtr' swig.cs(27333,68): error CS1525: Unexpected symbol `(' swig.cs(27334,45): error CS1525: Unexpected symbol `)' swig.cs(27334,59): error CS1514: Unexpected symbol `new', expecting `.' or `{' swig.cs(27334,57): error CS1530: Keyword `new' is not allowed on namespace elements swig.cs(27334,60): error CS1525: Unexpected symbol `switch_event' Compilation failed: 14 error(s), 0 warnings Opening swig.cs in my development environment shows that it is, indeed, using reserved names for methods - renaming 'event' to 'fsevent' stops the C# compiler from complaining and mod_managed builds. Thinking that I was just using old code, I checked the git tree, there's an updated swig.cs, but it also uses events and params keywords, and also fails to build. Does anyone know what I'm missing? Erik

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WebRTC no media sometimes
by Seven Du 16 Dec '13

16 Dec '13

Hi, So I findally got some time trying WebRTC. I noticed that some clients has no media when using WebRTC. I using the latest master with jssip 3.0. While my chrome(30.0.1599.101) has no problem all the time, other chromes elsewhere seems has no luck. e.g. 31.0.1650.63 and 33.0.1737.0 canary. I listed 3 channels below, looks like only my channel has 'AES_CM_128_HMAC_SHA1_80' and other channels has no that and no media. Also I see no auto stun for a long time from the log. Before trying more, is any hint it's a Chrome version problem or anything else? the no audio stun for a long time log has anything related to the client or only related to server side? Any configurable params? Thanks. L16,48000,768000,opus,48000,0,srtp:sdes:AES_CM_128_HMAC_SHA1_80,webRTC,1000@115.29.169.114,,ACTIVE,,,,,, L16,48000,768000,opus,48000,0,,webRTC,1000(a)115.29.169.114,,ACTIVE,,,,,, L16,48000,768000,opus,48000,0,,webRTC,1000(a)115.29.169.114,,ACTIVE,,,,,, 3 total. 2013-12-13 19:18:24.459945 [ERR] switch_rtp.c:718 No audio stun for a long time! 2013-12-13 19:18:35.479939 [ERR] switch_rtp.c:718 No audio stun for a long time! -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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Authenticated users still using public context
by Nathan Samson 16 Dec '13

16 Dec '13

Hi, I fear I am going to ask a very stupid and/or commong question, but after searching the internet I stil lcoudln't find the answer. My Setup * I have users in the directory (see an example configuration at the bottom of this mail) * Authenticated users should be able to use the default Dialplan * I want my DID provider to be able to make calls to me, it should use the public Dialplan / context. My DID provider is mydivert but that seems to be irrelevant. * I only have one SIP Profile (external) which has as default context public. All my users are connecting from the internet and not from the local network, so the internal profile would not make sense (or am I wrong?) My problem is that calls from nathan(a)sip.radarwerk.be are NOT using the default context, but are using the public context. 2013-12-15 15:04:53.206937 [DEBUG] switch_core_state_machine.c:117 sofia/external/nathan(a)sip.radarwerk.be Standard ROUTING "2013-12-15 15:04:53.206937 [INFO] mod_dialplan_xml.c:558 Processing nathan <nathan>->1999 in context public" The user nathan in the directory has user_context set to default, so I do not get why this is happening. See below for my configuration. [root@sip freeswitch]# cat directory/default/nathan.xml <include> <user id="nathan" number-alias="1200"> <params> <param name="password" value="**"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Nathan Samson"/> <variable name="effective_caller_id_number" value="1200"/> <variable name="outbound_caller_id_name" value="Nathan Samson"/> <variable name="outbound_caller_id_number" value="1200"/> </variables> </user> </include> Anyone an idea why the user_context is ignored? I am happy to provide my complete configuration. But basically I am using freeswitch-config-rayo 1.2.16 from the CentOS 6 repositories with some small modifications (mainly adding mydivert gateway, my users and my custom extensions in the default directory). Regards, Nathan

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Gateway Registration Issues
by Aloysius Lloyd 15 Dec '13

15 Dec '13

Hi All, I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I have the following gateway configuration <include> <gateway name="wwc_test"> <param name="auth-username" value="username"/> <param name="username" value="14161231234" /> <param name="password" value="password" /> <param name="proxy" value="sbc1.iristel.net" /> <param name="realm" value="iristel.net" /> <param name="expire-seconds" value="120"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="ping" value="25"/> </gateway> </include> The above configuration is not working. I have the following error in cli Error. *2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234 Registration Failed with status Conflict [409]. failure #1* ==== Also carrier saying there is conflict in the contact field.Here is how they receive now. *Contact: <sip:gw+test_iristel@10.20.30.1:5080 ;transport=udp;gw=test_iristel>.* But the contact field should be like below Contact: <14161231234@10.20.30.1:5080; Any suggestions? Thanks in advance. Lloyd

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