January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

T38 Configuration
by Robert Craig 21 Jan '13

21 Jan '13

Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! Robert

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FreeSWITCH Weekly News and Notes
by Michael Collins 21 Jan '13

21 Jan '13

Hello to our great OSS community! We are happy to report that the first ClueCon 2012 videos have been posted to our YouTube channel <http://www.youtube.com/cluecon>! The first talks posted are: * I'm Living In The Wrong Future!<http://www.youtube.com/watch?v=7SPpicsrbac>- Dr. Moshe Yudkowsky * ClueCon 2012 Keynote Address <http://www.youtube.com/watch?v=5bQansGixpI>- Anthony Minessale This week we will be posting a few more videos from day one of ClueCon 2012<http://www.cluecon.com/schedule/2012/>. Join this week's conference call or subscribe to our YouTube channel to find out which videos are uploaded next. Now that calendar year 2013 is in full swing we are making plans for ClueCon 2013, which will be held on August 6-8, 2013. Once again we'll be at the Hyatt Chicago Magnificent Mile <http://www.cluecon.com/hotels/2012/>. Stay tuned for more information about this year's event. The development team has been working hard on some cool new things which will be announced in due course. In the meantime Ken Rice is working very hard on getting FreeSWITCH 1.2.6 ready for final release. If you have a sandbox system that you can use to test the latest git HEAD please do so. Also, feel free to visit our bug tracker and see if there are any bugs that you can fix or verify as was discussed on last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_16>. This week we will be having an open discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_23>on topics that are dear to you. Please bring your questions for discussion and any tips and tricks that you'd like to share with the community. Have a great week! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Accessing SIP Replaces Header
by afshin afzali 21 Jan '13

21 Jan '13

Hi Guys, I'm receiving INVITE message which contains Replaces header inserted by Cisco gateway. I need to be able to read it from some channel variable but I could not find in any of the my channel variables! BEST, Afshin Afzali

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mod_xml_cdr and transfers - missing call legs
by Cal Leeming [Simplicity Media Ltd] 21 Jan '13

21 Jan '13

Hello, So, we are currently testing mod_xml_cdr and having some difficulties getting all the correct data. I actually have a fairly well commented dump of this happening (albeit from an unrelated bug); http://jira.freeswitch.org/secure/attachment/17781/debug-18-01-2013.txt Take the following scenario: >From user/2000 TO user/2001 from user/2001 to user/2002 via user/2000 from user/2002 to gw/447534971120 via user/2001 from gw/447534971120 to ivr/2003 via user/2001 from gw/447534971120 to gw/442477062086 via ivr/2003 HANGUP from gw/447534971120 At this point the CDRs are posted back to our web app (both A and B leg).. however, it doesn't seem to contain all the above call events.. The end goal for me was to have each stage of a call logged and linked.. so if a call is transferred 10 times, I will see 10 CDR legs. So, my questions are; * Is this behaviour expected? * Am I approaching this from the wrong angle? * Are there better ways to collect CDRs? Any and all feedback would be much appreciated. Thanks Cal

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mod_xml_curl - request for comments
by Cal Leeming [Simplicity Media Ltd] 21 Jan '13

21 Jan '13

Hello, I've been re-reading my changes to mod_xml_curl documentation, and I'm still not entirely happy with it. I can't tell if it genuinely needs a different approach, or if I'm just being fussy. Therefore, constructive criticism is very much welcome - even if it means the whole damn thing has to be re-done again. http://wiki.freeswitch.org/wiki/Mod_xml_curl Thanks Cal

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SBC In-band DTMF
by support＠ecn.net.au 21 Jan '13

21 Jan '13

Hi All We're quite new to Freeswitch and are in the process of migrating from OpenSer (as an SBC) to Freeswitch. Mostly all is working well, except an oddity on DTMF. Our scenario: Telco/SIP Provider A is passing us calls using DTMF inband. We have a freeswitch configured as a SBC using 2 sip profiles (telco and internal) to topology hide and manage distribution of calls to the PBX servers located behind the SBC. The freeswitch will be handling up to a few hundred calls so we're trying to keep it lightweight. Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling customer needs. An example inbound call profile looks like this: <extension name="Inbound 124356"> <condition field="destination_number" expression="^(123456)$"> <action application="pre_answer"/> <action application="start_dtmf" /> <action application="bridge" data="sofia<mailto:sofia/external/123456@INTERNAL.PBX.IP:5060%22/>/external/123456@INTERNAL.PBX.IP:5060"/<mailto:sofia/external/123456@INTERNAL.PBX.IP:5060"/>> </condition> </extension> Initially when calling into the platform IVR type applications runinng on our PBX boxes would not work (you could hear the DTMF but the platform did not recognise the tones). We have had to add the appliation start_dtmf in order for Freeswitch to pass the DTMF to the Asterisk PBX behind the SBC. Interestingly on our OpenSer platform we just proxied the media (rtpproxy) with inband DTMF from the Telco and our PBX boxes recognised the inband DTMF tones on the PBX platforms and IVR type applications just worked. However under freeswitch if we don't start_dtmf before the bridge the backend PBX boxes don't recognise the DTMF inband (even though the tones are audible ie you can hear them on a call recording on the PBX). Have we missed something here? We would have thought with inband DTMF on non compressed codec (no transcoding) that the tones would just work with the media stream? We have confirmed both legs are PCMA and when using start_dtmf the first second of the call is clipped. Kind Regards,

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FAX t38_gateway operation confusion (mod_spandsp)
by João Mesquita 20 Jan '13

20 Jan '13

Hello everyone! I have been playing around with mod_spandsp's capabilities lately and most certainly because I am new to the all old faxing stuff (pretty funny, right?), I have been making a bit of a mess in my head about how a T30->T38 gateway should work. I've set up a testing environment here as follows: *Components:* * * 2 analog fax machines (call is FaxA and FaxB) 2 FXS cards 2 FreeSWITCH machines (call FSA and FSB) *Connection scheme:* * * *FaxA <-> FSA and FaxB <-> FSB* Both of these connections are using the respective FXS cards *FSA <-SIP-> FSB* * * Both FreeSWITCHes are connected using SIP, of course. *Call scenarios:* 1. *FaxA sending a fax to FSB (using rxfax on FSB) - OK* Dialplan on FSA is: <action application="set" data="sip_execute_on_image="t38_gateway peer nocng" /> <action application="bridge" data="{fax_t38_enable=true}sofia/internal/FSB" /> Dialplan on FSB: <action applcation="answer"/> <action applcation="set" data="fax_t38_enable=true" /> <action applcation="set" data="fax_t38_request=true" /> <action applcation="rxfax" data="/tmp/fax.tif" /> 2. *FSB sending a a fax to FaxA - OK* Dialplan on FSA is: <action application="set" data="sip_execute_on_image="t38_gateway self nocng" /> <action application="bridge" data="{fax_t38_enable=true}sofia/internal/FSB" /> Originate line on FSB: originate {fax_enable_t38_request=true,fax_enable_t38=true}sofia/internal/FSB 'set:ignore_early_media=true,txfax:/root/fax.tiff' inline 3. *FaxA sending a fax to FaxB in manual mode* 4. *FaxA sending a fax to FaxB in automatic mode* 5. *FaxA receiving a fax from FaxB in manual mode* 6. *FaxA receiving a fax from FaxB in automatic mode* Here comes the problem. It is not clear to me how this should work from the dialplan perspective and from the theory as well. Now, as I understand, t38_gateway works by detecting CED tones as specified on the code by the line: *spandsp_fax_detect_session(session, "rw", timeout, MODEM_CONNECT_TONES_FAX_CED_OR_PREAMBLE, 1, direction, NULL, t38_gateway_start);* *Questions:* If this is true and looking at the T.30 standard, some of the cases mentioned above make the CED tone optional, is this correct? By talking to Steve Underwood yesterday on IRC, he mentioned that (and I quote): "the emitting gateway should detect the signal from the answering FAX machine and initiate the change to T.38". My understanding is that on 3 for example, FSA needs to detect CED being sent from FaxB, correct? Looking at the T.38 standard Annex D to understand who should initiate a T.38 re-invite at what instance, I found this phrase: > "The receiving gateway should only use the ITU-T V.21 preamble generated > by the answering FAX endpoint as the trigger for detection, and it should > generate the INVITE request no more than 200 milliseconds after the ITU-T > V.21 preamble begins, in order to allow the maximum amount of time for the > session to transition into ITU-T T.38 mode before the answering FAX > endpoint begins transmitting NSF, CSI or DIS." Googling, I found that the ITU-T V.21 preamble is not CNG or CED tone, but rather something else... I assume that the _OR_PREAMBLE is exactly that, correct? I greatly appreciate any help as the Wiki is not very specific on the analog fax stuff. My intent is to wikify all that and help the folks out there as well. Fax seems to be mistic for more than myself. Regards, João Mesquita

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bluetooth endpoint
by peter ness 20 Jan '13

20 Jan '13

Are they any current soultions that enable bluetooth as an endpoint.

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mod_voicemail always sends empty emails
by Charlie Orford 20 Jan '13

20 Jan '13

Hi list, Voice mails get correctly saved in /usr/local/freeswitch/storage/voicemail/default/domain.com/extnum/ but the email sent by mod_voicemail always arrives blank with the wav file attachment missing. To check the problem wasn't mail server related, I setup a simple postfix "Local only" installation on the same host as FS. Here's the entire contents of a voicemail email sent to postfix by FS (note: domain.com is substituted for real domain): # cat /var/mail/root From voicemail(a)domain.com Sat Jan 19 17:38:03 2013 Return-Path: <voicemail(a)domain.com> X-Original-To: postmaster(a)fs.domain.com Delivered-To: postmaster(a)fs.domain.com Received: by fs (Postfix, from userid 1001) id 9E6AC23E57; Sat, 19 Jan 2013 17:38:03 +0000 (UTC) Message-Id: <20130119183349.9E6AC23E57@fs> Date: Sat, 19 Jan 2013 18:33:49 +0000 (UTC) From: voicemail(a)angloeuropean.com To: undisclosed-recipients:; And here are the voicemail specific config params for the extension: <param name="vm-password" value="1234"/> <param name="vm-attach-file" value="true"/> <param name="vm-mailto" value="postmaster(a)fs.domain.com"/> <param name="vm-mailfrom" value="voicemail(a)domain.com"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-message-ext" value="wav"/> <param name="vm-keep-local-after-email" value="true"/> /usr/local/freeswitch/conf/voicemail.tpl exists and looks correct (contents here: http://pastebin.com/Tr4fvyht ) I took a log (via fsi_cli and /log 7) of a call leaving a voicemail to that extension (here: http://pastebin.com/B1zteGV7 ) but nothing jumps out. Is there anything else I can do to track down what's causing this? I'm running the latest stable branch on Debian squeeze: FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) Thanks, Charlie

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Digium TDM410 + DTMF + echo cancel
by Tim Bock 20 Jan '13

20 Jan '13

Hi, I am having a continuing problem with freeswitch and the Digium TDM card. For some period of time after a reboot, DTMF is detected correctly for incoming calls, and there is no problem with echo. All is good. Then at some point, dtmf is no longer detected on incoming calls handled by the TDM card. Presumably at the same time that dtmf quits being detected, the echo cancellation quits working. Again, just on incoming calls handled by the Digium card. The only way I've been able to get things working again is to reboot the server, but it doesn't seem to last for long. Can anyone suggest how I might track down what is wrong? Thanks, Tim

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