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Set command is not working with the variable value in freeswitch
by Nikhitha T 14 May '14

14 May '14

In freeswitch as I am setting the variable like this *<action application="set" data="RESULT=${system sed -n 1p /nfs/th/play/${uuid}}" inline="true"/>* Here i am trying to assign the wav file path to the variable,but here the problem is it is getting assigned as *2014-05-08 13:34:56.659049 [DEBUG] mod_dptools.c:1402 sofia/internal/1002(a)192.168.0.106 <1002(a)192.168.0.106> SET [RESULT]=[/usr/local/freeswitch/sounds/en/us/callie/directory/8000/dir-for_prev.wav]* The variable RESULT is not being set properly,The closing square bracket is coming to the new line as it should be in the same line immediately after the path.So while playing the wav file it is showing the error as *2014-05-08 13:34:56.679046 [ERR] switch_core_file.c:150 Invalid file format [wav] for [/usr/local/freeswitch/sounds/en/us/callie/directory/8000/dir-for_prev.wav]!* Can anyone help me in solving this issue.Thanks in advance -- Thanks & Regards --------------------------------- Nikhitha

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Problem with send_dtmf w.r.t bind_digit_action in Java ESL
by Nikhitha T 14 May '14

14 May '14

Hello, I am using bind_digit_action application in freeswitch dialplan,Once i receive the dtmf i'll transfer to some extension. This is working fine when the user press the dtmf but My requirement is to send the dtmf from java ESL , I used send_dtmf application but bind_digit_action didn't work. *Actions in dialplan* <action application="bind_digit_action" data="my_digits,2,exec:execute_extension,108 XML default,both,both"/> <action application="bind_digit_action" data="my_digits,1,exec:execute_extension,109 XML default,both,both"/> *Code in Java ESL* public void sendDtmf(Channel channel,int value) { SendMsg dtmfMsg = new SendMsg(); dtmfMsg.addCallCommand("execute"); dtmfMsg.addExecuteAppName("send_dtmf"); dtmfMsg.addExecuteAppArg(""+value); // dtmfMsg.addEventLock(); EslMessage response = sendSyncMultiLineCommand(channel, dtmfMsg.getMsgLines()); } Please help me with this issue..Thanks in Advance -- Thanks & Regards --------------------------------- Nikhitha

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Re: [Freeswitch-users] Is QoS needed / beneficial?
by Deon Vermeulen 14 May '14

14 May '14

Here is a pretty good guide for QoS. http://cisconerd.wordpress.com/2012/09/13/qos-cheatsheet/ And a Sample Cisco Template for CE devices: http://cisconerd.wordpress.com/2012/09/13/qos-template/ Any Operator knowing what they do should follow this concept on throughout their Core. It works when implemented properly and standards are adhered to throughout the network. Regards --------------- Original Message --------------- Date: 5/13/14, 10:47 PM From: Joli Martinez <mrjoli021(a)gmail.com> Subject: Re: [Freeswitch-users] Is QoS needed / beneficial? >>Look at MicroTik. They support QoS and are inexpensive. >> >>Sent from my iPhone >> >>> On May 13, 2014, at 10:24 AM, Francis <sms(a)icefire.qza.net.au> wrote: >>> >>> Yes. :) >>> >>> The long answer is, unless your provider's bandwith is _guaranteed_ to _always_ be greater than your total requirements, there will be potential bottlenecks as traffic from different nodes compete with one another. Bear in mind that one server downloading updates is enough to reduce inbound SIP calls to a jittery, garbled mess. Add to that the fact that most (all?) providers have no QoS policy at all, so you are reliant on on a router that is capable of bidirectional control, not just inbound. >>> >>> Having said that, do you know if the firewall or ADTRAN are running some kind of QoS policy already? You might not need to buy extra parts..... >>> >>> >>> >>> Francis >>> >>> >>> >>> >>>> On 2014-05-14 06:05, Jason Moran wrote: >>>> >>>> We currently plug our CentOS 5.x FreeSWITCH server directly into our service provider’s ADTRAN to connect with the outside world, and to a switch for intranet (of which a 2nd firewall is connected there and to the ADTRAN for our other servers). We are shopping for other providers who are shocked that we don’t have a router in between that is doing QoS to prioritize the voice/SIP traffic. >>>> >>>> >>>> Is a router w/ QoS capabilities a good idea? >>>> >>>> >>>> Thanks, >>>> Jason >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)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://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)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://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting(a)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://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >>

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Is QoS needed / beneficial?
by Jason Moran 13 May '14

13 May '14

We currently plug our CentOS 5.x FreeSWITCH server directly into our service provider's ADTRAN to connect with the outside world, and to a switch for intranet (of which a 2nd firewall is connected there and to the ADTRAN for our other servers). We are shopping for other providers who are shocked that we don't have a router in between that is doing QoS to prioritize the voice/SIP traffic. Is a router w/ QoS capabilities a good idea? Thanks, Jason

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Freeswitch Week in Review(Master Branch) May 4th-10th
by Kathleen King 13 May '14

13 May '14

Hello, again. Last week in the FreeSWITCH master branch we had 52 commits. The miscellaneous Coverity fixes continued with some minor code tweaks and removal of dead code. We saw some more support for GSM added in mod_freetdm, progress toward data modem support in spandsp, and Openzap was removed from tree because it has been replaced with freetdm. REMINDER: ClueCon 2014 is coming up fast! Make sure you register! The following bugs were squashed: 752f795 fix compiler error to nlsml.c from attempted Coverity fix in mod_rayo 0a2ba88 fixed bug where all notify sending are treated as though they are natted instead of only when nat is detected in mod_sofia 3ff3f32 added error response to prevent segfault reported by mod_managed Jira: http://jira.freeswitch.org/browse/FS-6484 2459e71 prevent the jitter buffer from over frequent resets in bad connection scenarios in libstfu Jira: http://jira.freeswitch.org/browse/FS-6337 4a543b2 fixed bug where ACL "domains" fails to parse CIDR of users located in domain section in FS core Jira: http://jira.freeswitch.org/browse/FS-6506 88ce7da minor tweak to make ws code work in blocking mode properly when used outside sofia New features that were added: 68a3250 added work in progress for data modem support b8a32ed adding support for GSM module a82e7a7 Added GSM debug configuration parameter in freetdm Improvements in cross platform build supports: c96c733 Test for pkg-config in configure 28ee773 put APR_LIBS back in LIBADD and LDFLAGS, but add some magic to .DEFAULT to ignore -l targets e49feac don't re-build tone2wav and fs_encode on every build aef982d added work around for Mac's broken sed 3c0e40e Remove bashism from our Makefile hack d6fe109 remove opezap from tree, it has been replaced by freetdm a058939 added patch to ensure that user specified libraries have a higher priority for openssl Jira: http://jira.freeswitch.org/browse/FS-6475 2508ae9 unset -l libs from apr-config as dependencies in the Makefile c856023 Fix missing $ for escaping shell variables for make 84eb4a0 re-added Makefile for iks helpers unit tests in mod_rayo 31069f3 set to generate the modules list vars from modules.conf into a file instead of on the fly every time 04dc901 Check for pkg-config at $PKG_CONFIG in configure d7d4625 re-added Makefile for srgs unit tests in mod_rayo f809d40 added quickie targets for FS sounds In terms of stability these were the use cases that were fixed: ee6f3da CID: 1211948 fix memory leak of decoded iks message in function parse_plain_auth_message in mod_rayo 717bcf6 fixed memory leak in switch_ivr_collect_digits_count Jira: http://jira.freeswitch.org/browse/FS-6515 8488694 fixed a leak in error case in libscgi 2a50bbb CID: 1211950 fix memory leak of iksparser in function nlsml_parse in mod_rayo Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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remove precondition
by Jason Holden 13 May '14

13 May '14

How do I remove precondition from the supported header in sip invites?

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New approach for integrating FreeSWITCH and HylaFax
by Markus Lindenberg 13 May '14

13 May '14

Hi, We just open sourced GOfax.IP, a project i've been working on for the better part of the last two months. It's a new (and I think best yet) way to bring "native" Fax over IP using T.38 or G711 to HylaFAX Fax servers by directly connecting HylaFAX to FreeSWITCH's mod_spandsp without any modem emulation layer. We replaced the faxgetty and faxsend processes with versions that use FreeSWITCH to handle incoming and outgoing calls. https://github.com/gonicus/gofaxip We wanted to have a solution that brings all the power of rxfax/tfax to HylaFax. FreeSWITCH's modem emulation has to be chained to t38_gateway to use T.38, and in my tests that didn't work as well as using txfax/rxfax, so I looked at how capi4hylafax hooks into HylaFax and implemented that approach for FreeSWITCH. I'd love to hear thoughts/suggestions and am happy to accept bug reports and contributions. Regards, Markus

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Routing for custom user
by Роман aka iTUX 13 May '14

13 May '14

Hello! I have several users in freeswitch/directory/default (for example 5001, 5002, 5003). Also have several ISP in freeswitch/sip_profiles (isp0, isp1, isp2, isp3). Default routing define in freeswitch/dialplan/default and by default used isp0 for outbound calls. How i can organize outbound calls for user 5001 via isp1 and for user 5002 via isp2? Or user 5001 and 5002 via isp1, but with different A-numbers (aon) Thanks for advice -- Kind regards, Roman

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Can't make calls with FXO port with FreeTDM
by neo haux 12 May '14

12 May '14

Hi all, I am struggling to get my TDM400P card works with FS FreeSWITCH Version 1.5.13b+git~20140510T065717Z~b8a32ed0f3~32bit (git b8a32ed 2014-05-10 06:57:17Z 32bit) on a Via mother board with "VIA Samuel 2" CPU I can receive call from PSTN (I am in Canada) via my FXO port with no problem. but impossible to make outgoing calls form internal station to PSTN(neither from SIP client nor from analog phone connected to FXS port) . it keeps telling me that the line is [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] but that's not true ! hte line is not busy I have the same hardware with asterisk and all work smoothly. I have no idea where to look for to debug this problem. My FTDM list shows: freeswitch@internal> ftdm list +OK span: 1 (myFXO1) type: analog physical_status: ok signaling_status: UP chan_count: 1 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options: none +OK span: 2 (myFXS3) type: analog physical_status: ok signaling_status: UP chan_count: 1 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options: call swap The interesting (strange) lines I read from the console when I start a call to PSTN: .... .... .... freeswitch@internal> 2014-05-12 10:57:05.857571 [DEBUG] ftmod_analog.c:503 [s1c1][1:1] Changed state from DIALING to BUSY 2014-05-12 10:57:05.877579 [DEBUG] ftmod_analog.c:638 [s1c1][1:1] Completed state change from DIALING to BUSY in 20 ms 2014-05-12 10:57:05.877579 [DEBUG] ftmod_analog.c:644 [s1c1][1:1] Executing state handler on 1:1 for BUSY 2014-05-12 10:57:05.877579 [DEBUG] ftmod_analog.c:803 [s1c1][1:1] Changed state from BUSY to DOWN 2014-05-12 10:57:05.897586 [DEBUG] ftmod_analog.c:638 [s1c1][1:1] Completed state change from BUSY to DOWN in 20 ms 2014-05-12 10:57:05.897586 [DEBUG] ftmod_analog.c:644 [s1c1][1:1] Executing state handler on 1:1 for DOWN 2014-05-12 10:57:05.897586 [DEBUG] mod_freetdm.c:2262 got FXO sig 1:1 [STOP] 2014-05-12 10:57:05.897586 [NOTICE] mod_freetdm.c:2282 Hangup FreeTDM/1:1/5141111111 [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] 2014-05-12 10:57:05.897586 [DEBUG] switch_channel.c:3216 Send signal FreeTDM/1:1/5141111111 [KILL] ... ... ... I don't want to past here a big page of debug output. The complete output can be read here: http://pastebin.freeswitch.org/22524 Any idea or suggestion is highly welcome. Thanks.

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Using XML CURL for LRN queries
by Victor Chukalovskiy 12 May '14

12 May '14

Good day, I have a web server that I can request as following: http://192.168.1.2/pqrs/route?tn=4165555555 And it will respond with xml containing LRN which looks like this: <pqrsd> <api>pqrsd</api> <operation>route</operation> <status>success</status> <tn>4165555555</tn> <region>canada</region> * <rn>4160000001</rn>* <spid/> <spName/> <routedBy>none</routedBy> </pqrsd> Can I use mod_xml_curl to make LRN queries from XML dialplan and put LRN from the response into a channel variable? Is it suitable for such real-time queries? If so, how should I setup mod_xml_curl? WiKi mostly goes over standard stuff such as getting directory, dialplan, ACL etc Thank you, -Victor

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