December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

t.38 Call Flow Information
by Spencer Thomason 22 Dec '12

22 Dec '12

Hello, I've been analyzing some Wireshark traces to get a better grasp on the t.38 gateway process. I'm using t38_gateway to convert SIP t38 to TDM audio on the receiving end. I'm struggling to understand the following call flow. I've excluded the audio portion for brevity but t38 is negotiated correctly by both endpoints (FS on our end and Acme on the other). Note the absence of a DIS message and the PPS prior to CFR. Can anyone shed any light on this? Thanks in advance, Spencer Acme FreeSWITCH |1716.691635| INVITE SDP (t38) | |SIP Request | |(5060) <------------------ (5070) | | |1716.719556| 100 trying -- your call is important to us | |SIP Status | |(5060) ------------------> (5070) | | |1716.830431| | no-signal | |t38:t30 Ind:no-signal | | |(21440) <------------------ (15876) | |1716.906132| 200 OK SDP (t38) | |SIP Status | |(5060) ------------------> (5070) | | |1716.912839| ACK | | |SIP Request | |(5060) <------------------ (5070) | | |1716.973011| | no-signal | |t38:t30 Ind:no-signal | | |(21440) ------------------> (15876) | |1717.450595| | v21-preamble |t38:t30 Ind:v21-preamble | | |(21440) <------------------ (15876) | |1717.730869| | PPS | |t38:v21:HDLC:Partial Page Signal | | |(21440) <------------------ (15876) | |1717.780811| | no-signal | |t38:t30 Ind:no-signal | | |(21440) <------------------ (15876) | |1718.130798| | v17-14400-long-training |t38:t30 Ind:v17-14400-long-training | | |(21440) <------------------ (15876) | |1719.541095| | t4-non-ecm-data:v17-14400 |t38:t4-non-ecm-data:v17-14400 Duration: 1.51s No packet lost | | |(21440) <------------------ (15876) | |1721.060850| | no-signal | |t38:t30 Ind:no-signal | | |(21440) <------------------ (15876) | |1722.573572| | v21-preamble |t38:t30 Ind:v21-preamble | | |(21440) ------------------> (15876) | |1723.593652| | CFR | |t38:v21:HDLC:Confirmation To Receive

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Re: [Freeswitch-users] core dump when compiling master git
by Peter Olsson 21 Dec '12

21 Dec '12

One common issue is that you've set "ulimit -s 240" to start FS, and then it's active in the running shell. Try to use login into new session and see if it helps. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale Skickat: den 21 december 2012 01:34 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] core dump when compiling master git It looks like a seg in the helper app that generates part of the code in the spandsp build so I can see it being a real bug but really it may be a bug in a compliler or something too since we don't all suffer from this same problem. It must be partially environmental. +10 points for the form letter!!!! w00t JIRA FTW! Also we brought back "make sure" by popular demand that nukes the whole tree and recompiles. On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice(a)freeswitch.org<mailto:krice@freeswitch.org>> wrote: No its not even in spandsp... This happens when he is building spandsp... This means something like gcc is segfaulting... On 12/20/12 5:22 PM, "Brian Foster" <bdfoster(a)endigotech.com<http://bdfoster@endigotech.com>> wrote: Technically, yes, SpanDSP but I thought Steve floats around JIRA? On Thu, Dec 20, 2012 at 6:16 PM, Ken Rice <krice(a)freeswitch.org<http://krice@freeswitch.org>> wrote: This is not a Segfault in FreeSWITCH you both fail.... This is probably a build system issue.... I would recommend pastebinning and a full trace from a clean tree of the bootstrap, configure and build process... On 12/20/12 5:06 PM, "Michael Collins" <msc(a)freeswitch.org<http://msc@freeswitch.org> <http://msc@freeswitch.org> > wrote: Brian, have you typed this one before? :D Juan, yes, definitely file a Jira on this. Also, collect the backtrace on that segfault if you can and attach as a txt file to the Jira case. -MC On Thu, Dec 20, 2012 at 2:59 PM, Brian Foster <bdfoster(a)endigotech.com<http://bdfoster@endigotech.com> <http://bdfoster@endigotech.com> > wrote: Dear OP, We've detected that this is a potential bug in FreeSWITCH. Because this is a potential bug, the mailing list is really not the place to report this. Before you file a bug report, and if you haven't done this already, please make sure that the symptoms you are experiencing do not concur with another previously submitted bug report. Search JIRA <http://jira.freeswitch.org> to ensure this. You might want to check the Wiki just in case this is a feature, not a bug. You can make a difference. Bugs can be reported to the FreeSWITCH JIRA <http://jira.freeswitch.org> . By doing this one simple task, you can save the FreeSWITCH developers the blood, sweat, and tears that are needed in order to gather information related to your bug report and track the commits needed to fix the problem. By filing a JIRA, you also help the developers gather relevant information so they can take the best course of action. Developers ask for several pieces of information when filing a report. Take a look at the Reporting Bugs <http://wiki.freeswitch.org/wiki/Reporting_Bugs> page for more details and instructions. The more information you can gather, the easier it will be to fix the issue and the faster a commit can be made. Remember that FreeSWITCH is an open source project. FreeSWITCH Developers do not get paid to fix bugs, they simply do it for the sake of the community. They sacrifice sleep, family time, brain cells, and Advil to help YOU. Show them that you care. Report bugs the correct way. On Thu, Dec 20, 2012 at 5:41 PM, Juan Pablo L. <jpablolorenzetti(a)hotmail.com<http://jpablolorenzetti@hotmail.com> <http://jpablolorenzetti@hotmail.com> > wrote: Hi, i m trying to update my development server (VM) by making a clean installation from git master, but i m getting a core dump while compiling: /bin/sh: line 1: 9386 Segmentation fault (core dumped) ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.h make[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139 make[6]: *** [all] Error 2 make[5]: *** [all-recursive] Error 1 make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la<http://libspandsp.la> <http://libspandsp.la> <http://libspandsp.la> ] Error 2 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 i followed the classic steps (bootstrap,configure and make) is anyone experimenting the same issues ??? my info is: Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686 Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux thanks! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<http://consulting@freeswitch.org> <http://consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<http://FreeSWITCH-users@lists.freeswitch.org> <http://FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net<http://irc.freenode.net> #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 !DSPAM:50d3ac3432766009014104!

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Cisco-signed SSL Certificates/Alternative phone suggestions
by Andrew Cassidy 21 Dec '12

21 Dec '12

Hi guys, just looking for a little help. I've been using Cisco SPA50x phones for a little while now, they have some pretty nice features for the price, including the ability to write custom applications in XML. However, to run them over HTTPS instead of HTTP, you need a certificate signed by Cisco, which I'm having difficulty obtaining. So my questions are: 1. Can anyone help me obtain such a certificate? I have the CSR already to go. 2. Are there any similarly priced and featured phones available that you guys have experience with? Thanks guys, -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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Tracing
by Miha 21 Dec '12

21 Dec '12

Hi, I am experiencing some problems related with media with some users. It is almost impossible to figure it out where is a problem because this problem is not presented all day. I need some sip trace dump for a specific user if it is possible due to a lot of traffic. It would be nice that this would not cause to much CPU load and disk consumption as this would need to be running and tracing 24/7. Users are reporting the thy exeriacing noise in conversation and sometime thy do not hear other side. What would be the best way? Thanks! Miha

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build error with new Freeswitch 'nametab.h': No such file or directory
by Alexander Haugg 21 Dec '12

21 Dec '12

Hi, i hope the question is not reapplied today i checkout the newest version of freeswqitch and get following error: "E:\TFS_neu\C4B UC\Main\3rd Party\FreeSWITCH\tmp_3\freeswitch\Freeswitch.2008.sln" (rebuild target) (1) -> (_Libraries\xmlrpc-c\xmltok:Rebuild target) -> ..\..\xmlrpc-c\lib\expat\xmltok\xmltok.c(10): fatal error C1083: Cannot open include file: 'nametab.h': No such file or directory Exist a workaround or bugfix for this problem? Thanks for your help

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core dump when compiling master git
by Juan Pablo L. 21 Dec '12

21 Dec '12

Hi, i m trying to update my development server (VM) by making a clean installation from git master, but i m getting a core dump while compiling: /bin/sh: line 1: 9386 Segmentation fault (core dumped) ./make_modem_filter -m V.17 -i -r > v17_v32bis_rx_fixed_rrc.hmake[7]: *** [v17_v32bis_rx_fixed_rrc.h] Error 139make[6]: *** [all] Error 2make[5]: *** [all-recursive] Error 1make[4]: *** [/home/fs/freeswitch/libs/spandsp/src/libspandsp.la] Error 2make[3]: *** [mod_spandsp-all] Error 1make[2]: *** [all-recursive] Error 1make[1]: *** [all-recursive] Error 1make: *** [all] Error 2 i followed the classic steps (bootstrap,configure and make) is anyone experimenting the same issues ???my info is:Linux pbxvm 3.1.5-1-ARCH #1 SMP PREEMPT Sun Dec 11 06:26:14 UTC 2011 i686 Intel(R) Core(TM) i5-2410M CPU @ 2.30GHz GenuineIntel GNU/Linux thanks!

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call_timeout fails for callcenter agents
by Juan Pablo L. 20 Dec '12

20 Dec '12

Hi, i m trying to configure a callcenter, i need the application to ring the agentsfor 10 seconds and then move on to the next agent in the list, everything works ok except that the agents ring for more than 10 seconds, so the channel var call_timeout is not being used bythe callcenter application, probably due to a misconfiguration somewhere but i cant see where ... this is my config for the agents: <agent name="2@callcenter" type="callback" contact=" ignore_early_media=true,originate_timeout=10,call_timeout=10]sofia/gateway/huawei_csoft/6611290" status="Available" max-no-answer="3" wrap-up-time="10" reject- delay-time="5" busy-delay-time="5" no-answer-delay-time="5" /> and this is my config in the dialplan (also tried with exports): <action application="set" data="hangup_after_bridge=true"/><action application="set" data="call_timeout=10"/><action application="set" data="originate_timeout=10"/><action application="set" data="ignore_early_media=true"/><action application="callcenter" data="callcenter@default"/> i can not see what i m doing wrong so i added that variables in both places (dialplan and the agents defs)and i see in the logs that they are being set in both places: EXECUTE sofia/external-huawei_gw/2802018@10.49.0.2:5062 set(hangup_after_bridge=true)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018@10.49.0.2:5062 SET [hangup_after_bridge]=[true]EXECUTE sofia/external-huawei_gw/2802018@10.49.0.2:5062 set(call_timeout=10)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018@10.49.0.2:5062 SET [call_timeout]=[10]EXECUTE sofia/external-huawei_gw/2802018@10.49.0.2:5062 set(originate_timeout=10)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018@10.49.0.2:5062 SET [originate_timeout]=[10]EXECUTE sofia/external-huawei_gw/2802018@10.49.0.2:5062 set(ignore_early_media=true)2012-12-19 15:38:16.716876 [DEBUG] mod_dptools.c:1335 sofia/external-huawei_gw/2802018@10.49.0.2:5062 SET [ignore_early_media]=[true]EXECUTE sofia/external-huawei_gw/2802018@10.49.0.2:5062 callcenter(callcenter@default)2012-12-19 15:38:16.716876 [DEBUG] mod_callcenter.c:2508 Member 2802018 <2802018> joining queue callcenter@default2012-12-19 15:38:16.736878 [DEBUG] mod_local_stream.c:417 Opening Stream [moh/8000] 8000hz2012-12-19 15:38:16.736878 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 20ms2012-12-19 15:38:16.776878 [DEBUG] mod_callcenter.c:1049 Updated Agent 2@callcenter set state = Receiving2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables2012-12-19 15:38:16.786922 [DEBUG] switch_ivr_originate.c:2425 Parsing session specific variables2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [ignore_early_media]=[true]2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [originate_timeout]=[10]2012-12-19 15:38:16.786922 [DEBUG] switch_event.c:1569 Parsing variable [call_timeout]=[10]2012-12-19 15:38:16.786922 [NOTICE] switch_channel.c:951 New Channel sofia/external-huawei_gw/6611290 [09d48fc6-d133-4673-9f12-074af601f1f2] kindly give me a hint what i m doing wrong, thanks a lot for your help.

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Yet another NAT question
by Sean Devoy 20 Dec '12

20 Dec '12

Hi, Since several people have suggested to avoid SIP ALG at all costs, I have disabled it and banged my head some more! Without SIP ALG, only one of my Polycom 335s can Register. The other 5 fail. (With SIP ALG they all register fine). Now I have found that changing: SETTINGS>SIP>LOCAL SIP PORT> to a UNIQUE port (5066 in this case) And SETTINGS>NETWORK>NAT>SIGNALING PORT> to the same UNIQUE port I got a second phone to REGISTER and exchange calls! When I say unique port, I mean unique to all the phones on this NAT Router. My question is "Is that fine?" I just want to know if there are limitations on the SIP Port number? Are there issues I should watch out for. Thanks, Sean

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Codec usage between FS and Gateway
by Ryan Watkins 20 Dec '12

20 Dec '12

Hello all! Got a question that I hope someone can help with… I'm trying to test Codec2 usage (mod_codec2) however, since there doesn't seem to be any hard/softphones out that use the codec, I've setup two FS servers in an attempt to do the following: SIP Phone < -- any codec --> FS Server 1 < -- codec2 --> FS Server 2 < -- any codec --> SIP Trunk < -- --> endpoint So my question is how do you enforce the use of mod_codec2 between the two FS servers, while still allowing the endpoints to use any codec they support? Thanks! Ryan Watkins Networking & Customer Support OCENS 22608 Marine View Drive South Suite 300 Des Moines, WA 98198 Satellite Systems and Services for Iridium, Inmarsat, Globalstar, KVH ________________________________________________________ www.ocens.com<http://www.ocens.com> | support.ocens.com<http://support.ocens.com> Office: (206) 878-8270 x1004 | Cell: (360) 521-7334 | Fax: (206) 878-8314

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[CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
by Sean Devoy 20 Dec '12

20 Dec '12

HI, I have a bridge statement used dozens and dozens of times a day: EXECUTE sofia/external_noauth/anonymous(a)64.136.174.30 bridge(user/300(a)fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/301(a)fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/203(a)fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/302(a)fs_mbri2.bizfocused.com :_: [leg_delay_start=8]user/303(a)fs_mbri2.bizfocused.com) Very occasionally, recently extensions (at least 300 and 302) have returned [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] All of these extensions are at the same remote location using the same wan links/routers, etc. What does that mean? Pastebin: http://pastebin.freeswitch.org/pastebin.php?dl=20306 Also, the 8 second delay appears to be ignored (I think, I have never asked the customer or been on site to test). Any ideas? Thanks, Sean

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