July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available!
by Phil Quesinberry 23 Jul '12

23 Jul '12

Thanks Anthony, I actually was about to file a Jira on this since my RTP port range is set from 16384 to 32768. I asked about the problem on the list before doing so because I wanted to see if anyone knew about it, since I had searched on Jira FIRST and found nothing. I think I can speak for everyone in saying that we all appreciate the hard work that goes into FS. For my part, I'll file a Jira in the future and follow-up on the mailing list if it looks like a bug. Regards, - Phil _____________________________________________ Anthony Minessale Mon Jul 23 20:37:09 MSD 2012 * Previous message: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/086041.htm l> * Next message: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/086053.htm l> * Messages sorted by: [ date ] <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/date.html> [ thread ] <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/thread.htm l> [ subject ] <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/subject.ht ml> [ author ] <http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/author.htm l> commit 524468be7ba5ee0887d2e975e4f131171c601a54 Author: Anthony Minessale Date: Mon Jul 23 11:36:19 2012 -0500 FS-4373 --resolve this is probably the same issue nobody would file a bug about on the mailing list even after I begged them to from last weekend please please stop filing bugs on the mailing list. I will still work on them and try to fix them but you are making me do way more work. Unless everyone wants to pay for support I cannot keep this up. _____________________________________________ From: Phil Quesinberry Sent: Sunday, July 22, 2012 1:16 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: [CRIT] sofia_glue.c:1083 No RTP ports available! Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet: sofia_glue.c:1083 No RTP ports available! Restarting FS fixes the problem. Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn't happening until the last week or two. Current version we're running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z Here's a quick snip from the console when attempting to dial out to show the context of the error: 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available! 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK] 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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make mod_rad_auth under FreeBSD 8.2
by Nikolay Rogoshchenkov 23 Jul '12

23 Jul '12

Hello, May be some-who knows how to compile mod_rad_auto under FreeBSD? This is what I'm got: =================================================================================================================================================== *dp# cd /usr/src/freeswitch/src/mod/applications/mod_rad_auth/* *You have new mail.* *dp# ./configure* *./configure: Command not found.* *dp# ls* *Makefile mod_rad_auth.c* *dp# make* *Compiling /usr/src/freeswitch/src/mod/applications/mod_rad_auth/mod_rad_auth.c...* *quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -I/usr/local/include -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/applications/mod_rad_auth/mod_rad_auth.c -fPIC -DPIC -o .libs/mod_rad_auth.o* *In file included from /usr/src/freeswitch/src/mod/applications/mod_rad_auth/mod_rad_auth.c:34:* */usr/local/include/freeradius-client.h:23:20: error: ISO C does not permit named variadic macros* **** Error code 1* * * *Stop in /usr/src/freeswitch/src/mod/applications/mod_rad_auth.* **** Error code 1* * * *Stop in /usr/src/freeswitch/src/mod/applications/mod_rad_auth.* *dp# * =================================================================================================================================================== Used git repo with last commit at Wed May 16 20:35:15 2012.

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non-t38 fax fails between two local FS servers
by James 23 Jul '12

23 Jul '12

I have two FS servers on the same LAN, one for receiving FAXIN and one for transmitting FAXOUT FAXOUT receives an originate via ESL to send a number that routes to FAXIN via a local FS GW. FAXIN directs call to python script via dialplan FAXIN drops call 10 secs after negotiation result with this error mod_spandsp_fax.c:500 Fax processing not successful - result (13) Unexpected message received. When I enable fax_enable_t38 and fax_enable_t38_request, this same call succeeds. This seems like bad configuration on FAXIN, but I receive some faxes from remote destinations without t38. -- James Lyons QuikVoIP, LLC. p: +1 (786) 369-5308 w: www.quikvoip.net

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Bria iOS does SRTP wrong.
by Brian West 23 Jul '12

23 Jul '12

Dear FreeSWITCHologists, Anyone that uses Bria on iOS, Please open a support case with CounterPath. I've tried, and they need to not offer a=crypto in an RTP/AVP and offer two m= lines one for RTP/AVP and RTP/SAVP in order of preference. They seem to think its not a critical thing to do SRTP the right way. (Guessing its not just SRTP LULZ!) Anyone see the refer to rfc3711 CRIT message? Why do I care so much... lolz -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410

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ZRTP - SAS-strings not the same
by "Finn Höck" 23 Jul '12

23 Jul '12

Hi, I got freeswitch working perfectly on CentOS 6.3 by following the guide on http://wiki.freeswitch.org/wiki/Installation_Guide. Then I followed the steps listet at http://wiki.freeswitch.org/wiki/ZRTP to get ZRTP working. Now, when I do calls from a zrtp-enabled client to another, communication still works BUT the sas-strings showing up on the clients aren't the same. Now my question is: why? Does anyone have some troubleshooting-tips for me? What possible reasons are there? Thanks in advance for any help. Greetings, Finn

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TLS and connection reuse
by Tihomir Culjaga 23 Jul '12

23 Jul '12

hello, im just wondering if it is possible to make FS re-use the existing TLS connection established on UA registratioin for incoming calls to UA? as an example... i have extensions 1002 (uses TLS) and 1009 (uses SIP/UDP). Both extensions are registered to the same FS. I can place calls from 1002 everywhere but 1002 cannot get any calls. http://pastebin.freeswitch.org/19575 recv 394 bytes from tls/[109.227.38.121]:60591 at 18:16:03.019253: ------------------------------------------------------------------------ REGISTER sip:85.114.35.241 SIP/2.0 Via: SIP/2.0/TLS 109.227.38.121:5061;rport;branch=z9hG4bK419550279 From: <sip:1002@85.114.35.241>;tag=376877386 To: <sip:1002@85.114.35.241> Call-ID: 665211288 CSeq: 1 REGISTER Contact: <sip:1002@109.227.38.121:5061 ;transport=tls;line=d85e74961f65cc6> Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 2000 Content-Length: 0 ------------------------------------------------------------------------ I know this register message is broken... good for TCP/UDP but bad for TLS :=) anyhow, extension 1002 establishes a TLS flow 109.227.38.121:60591 <> 85.114.35.241:5061 of course you know what happens when 1009 calls 1002, since 1002 advertized address:port in contact header different than the source port of the existing flow, FS tries to establish a new transport towards the UA and it fails. :=) So is there any way we can force FS to re-use the existing flow and send subsequent request messages via existing connection rather than trying to establish a new one that is going to fail miserably? The more i write this e-mail, the more i realize its a UA problem... don't advertize something you cannot get a call on... but anyhow... needed to ask that question :=) so, how do we fix this ? is the implementation of RFC5626 on client side the answer for that... does FS support this rfc at all ? the other solution should be to tie contact port to transport port but that's a hack.. any good advice in how to fix the UA behavior so it can send and receive calls via TLS connections? regards, Tihomir.

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FreeSWITCH Weekly News and Notes
by Michael Collins 23 Jul '12

23 Jul '12

Hello all! Last week we enjoyed a nice conference call with Darren Schreiber<http://www.cluecon.com/speaker/98?cc12-0723>from 2600hz <http://www.2600hz.com?cc12-0723>. Darren gave us a follow up<http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_18?cc12-0723>presentation to the original SIP 101 discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_20?cc12-0723>that was held in June. This was a great presentation that went beyond the basics of SIP. We discussed important concepts like Via and Record Route headers and NOTIFY messages. SIP can do a lot, and with these two presentations we really have only scratched the surface. If we all bug Darren enough then perhaps we'll get a third presentation! The audio of the presentation is in the usual place<http://torrents.freeswitch.org/conf_call_2012-07-18.torrent?cc12-0723>and the slides will be available shortly. This week we will be hearing from the community. We will have a brief report from Dave Kompel who will get us up to speed on using Windows Advanced Firewall to implement Fail2Ban<http://wiki.freeswitch.org/wiki/Fail2ban?cc12-0723>-like functionality. After Dave speaks we will be opening things up for community discussion. Please bring your questions and topics! ClueCon 2012 <http://www.cluecon.com?cc12-0723> is shaping up nicely! We have recently added Ditech Networks<http://www.ditechnetworks.com?cc12-0723> , Bandwidth.com <http://www.bandwidth.com?cc12-0723>, and Yealink<http://www.yealink.com?cc12-0723>as sponsors. We are also happy to announce that Darrell Hensley <http://www.youtube.com/watch?v=sLBLCcSiAr0> will be speaking on the subject of Transitioning to Professional Voice<http://www.cluecon.com/presentation/transition_pro_voice?cc12-0723>. Darrell is the CEO of GMVoices <http://www.gmvoices.com?cc12-0723>, a ClueCon media sponsor for many years and the supplier of Callie<http://www.youtube.com/watch?v=IfkRNgM1IiA>, the voice of FreeSWITCH. We look forward to seeing Darrell in person again. As a quick reminder, if you register<http://www.cluecon.com/register?cc12-0723>by the end of the day on July 25th you will still receive four chances to win in the great ClueCon giveaway. See you in two weeks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0723

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[CRIT] sofia_glue.c:1083 No RTP ports available!
by Phil Quesinberry 23 Jul '12

23 Jul '12

Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet: sofia_glue.c:1083 No RTP ports available! Restarting FS fixes the problem. Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn't happening until the last week or two. Current version we're running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z Here's a quick snip from the console when attempting to dial out to show the context of the error: 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available! 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK] 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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RTMP user not registered error
by Germán Ruiz 23 Jul '12

23 Jul '12

Hi. I 'm new with FreeSwitch and I 'm trying to make work the RTMP module in both directions. Now, I can do a call from an user registered in the flash application (using the example client) to an user connected with a SIP client (desktop or android application). But when I call from a SIP client to another user connected with RTMP, get error: mod_dptools.c:2960 Originate Failed. Cause: USER_NOT_REGISTERED I used the configuration example for RTMP (autoload_configs/rtmp.conf.xml). In the dialplan configuration, the bridge option applied is: <action application="bridge" data="user/${dialed}@my_domain"/> ¿What is missing to register the users connected with RTMP? Thanks and sorry for my bad english. Germán

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A fatal error has been detected by the Java Runtime Environment:
by bawajan 23 Jul '12

23 Jul '12

Hi, Am using ESL inbound connection to make calls and have a below flow (written in java) a) originating 15 calls simultaneously with park function b) play audio file c) creating new thread and passing eslconnection and playing a IVR in originate call. Here while handling events getting the below error : # A fatal error has been detected by the Java Runtime Environment: INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # SIGSEGV (0xb) at pc=0x00007fc01f1d3943, pid=14759, tid=140463084390144 INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # JRE version: 6.0_25-b06 INFO | jvm 3 | 2012/07/17 17:54:36 | # Java VM: Java HotSpot(TM) 64-Bit Server VM (20.0-b11 mixed mode linux-amd64 compressed oops) INFO | jvm 3 | 2012/07/17 17:54:36 | # Problematic frame: INFO | jvm 3 | 2012/07/17 17:54:36 | # C [libesljni.so+0xd943] long double+0x183 INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # An error report file with more information is saved as: INFO | jvm 3 | 2012/07/17 17:54:36 | # /usr/local/freeswitch/hs_err_pid14759.log INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # If you would like to submit a bug report, please visit: INFO | jvm 3 | 2012/07/17 17:54:36 | # http://java.sun.com/webapps/bugreport/crash.jsp INFO | jvm 3 | 2012/07/17 17:54:36 | # The crash happened outside the Java Virtual Machine in native code. INFO | jvm 3 | 2012/07/17 17:54:36 | # See problematic frame for where to report the bug. plz let me know, where am doing mistake and how to resolve it. Thanks in advance. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/A-fatal-error-has-been-detect… Sent from the freeswitch-users mailing list archive at Nabble.com.

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